| 00:00.44 | Kobaz | i'm having some weird sip/rtp problems... http://pastebin.com/m3b922470 |
| 00:04.55 | *** join/#asterisk jblack (n=jblack@pool-71-173-9-11.sctnpa.east.verizon.net) |
| 00:07.31 | *** join/#asterisk desdesdesdes (n=kg@196.211.34.2) |
| 00:07.35 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 00:18.19 | *** join/#asterisk mandd (n=moo@67.204.24.118) |
| 00:24.34 | *** join/#asterisk octothorp (n=vandebo@abydos.nerdbox.net) |
| 00:27.59 | octothorp | PickupChan isn't behaving the way I expect. I get "No channel found...", anyone around to help? |
| 00:38.08 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 00:41.22 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 00:45.24 | *** join/#asterisk paulproteus (i=paulprot@2002:db69:2513:0:0:0:0:1) |
| 00:46.32 | [TK]D-Fender | Octothorp : jsut show what you've got and see who answer |
| 00:53.18 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 00:54.27 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-8ee3a8e2ec3a2862) |
| 00:55.17 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 00:56.27 | octothorp | the guts of what I'm trying to do: http://nerdbox.net/pickup.conf |
| 00:56.42 | *** join/#asterisk jcoffi (n=jcoffi@75.147.155.89) |
| 00:57.28 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 00:57.48 | [TK]D-Fender | Octothorp :PB the failed attempt at verbose 10 |
| 00:59.44 | *** join/#asterisk HydroUK (n=hydrouk@unaffiliated/hydrouk) |
| 01:00.33 | desdesdesdes | can anyone please explain relaxdtmf=yes/no ? |
| 01:00.41 | *** join/#asterisk avrPhreaque (i=Default@dsl-dynamic-209-50-30-164.inebraska.com) |
| 01:00.43 | avrPhreaque | Hello |
| 01:01.31 | avrPhreaque | I am in need of an IC that acts as an interface between a host processor and a PSTN telephone instrument, with the host appearing as the CO. What do I need to look for? Camn that be had in a single package? |
| 01:01.48 | [TK]D-Fender | desdesdesdes: Whats to explain? I basical lowers the detection threshhold for noisy lines, and poor generators |
| 01:02.11 | [TK]D-Fender | avrPhreaque: What are you trying to make exactly? |
| 01:04.02 | avrPhreaque | [TK]D-Fender: I'd rather not say, exceopt to say that a standard telephone should be e\ab.e to p,ug into it, and think it is connected to the PSTN CO |
| 01:04.45 | jaytee | because it's obviously illegal |
| 01:04.50 | [TK]D-Fender | avrPhreaque: Plenty of other guides out there for this. You know the telephone ahsn't changed much in the past .... oh I don't know... CENTURY or so |
| 01:04.54 | desdesdesdes | i need to increase the waiting time for digits when transfering a call from fxs to fxs, because i am using cordless phones and when i press a number teh dtmf send there is a delay on it |
| 01:05.11 | octothorp | log at http://nerdbox.net/pickup.log |
| 01:05.16 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 01:05.19 | jaytee | and no one's going to suspect someone with Phreaque in their nick as being a telephone hacker. |
| 01:07.08 | [TK]D-Fender | Octothorp :that looks rather truncated |
| 01:07.30 | avrPhreaque | jaytee: there is absolutely nothing illegal about it. Every VoiP adpater on the market does it |
| 01:07.31 | octothorp | that's the entire log at verbose 10 of my test case |
| 01:08.13 | avrPhreaque | I am attempting to design a hardware module for VoiP like applications, I am not willing to say more than that, don't want to give our idea away |
| 01:08.36 | [TK]D-Fender | avrPhreaque: There are basic books and speca available everywere for this |
| 01:08.43 | jaytee | avrPhreaque, then all you need to do is reverse engineer a VoiP "adpater", whatever the hell that is :-) |
| 01:09.31 | [TK]D-Fender | Octothorp : For one thing, that channel isn't ringing. |
| 01:09.33 | jaytee | google You-Do-It Electronics |
| 01:09.49 | [TK]D-Fender | Octothorp: Next try it with a channel dump in-between |
| 01:10.12 | octothorp | i.e. show channels ? |
| 01:10.28 | [TK]D-Fender | Octothorp : show channels concise |
| 01:11.04 | octothorp | are there multiple notions of ringing? there's a "exten => s,n,Ringing" command on the channel before the pickupchan |
| 01:11.18 | avrPhreaque | jaytee: you're kidding, right? Any device that connects to the Network (TCP/IP-Internets) on one side, and has a jack for a PSSTN phone on the other side is a VoiP adapter |
| 01:11.37 | jaytee | ah! adapter! so that's what you meant. |
| 01:11.45 | avrPhreaque | I don't have ther $$$ to purchase one and tear it apart |
| 01:11.52 | jaytee | like an ATA, right? |
| 01:12.29 | avrPhreaque | ATA? I am unfamiliar with that acronym |
| 01:12.40 | x86 | analog telephone adapter |
| 01:12.42 | jaytee | don't have 29 bucks for a HandyTone 286? cripes, are you in the wrong room and wrong end of town. |
| 01:12.48 | [TK]D-Fender | Octothorp have your script do : System(asterisk -rx "core show channel thechannelinhere") so you can see in CLI live |
| 01:13.18 | [TK]D-Fender | avrPhreaque: 0 budget, 0 clue, and a "mystery plan". Well good luck with that |
| 01:13.48 | [TK]D-Fender | jaytee: Dodge is grossly overpopulated ;) |
| 01:13.49 | avrPhreaque | jaytee: I am a disabled hobbyist working with a programmer to develo a product idea. I have no money at all. Never heard of a HandyTone 286 |
| 01:13.51 | jaytee | goes off in search of parts to build an interociter |
| 01:15.18 | jaytee | avrPhreaque, the HandyTone286 is one of the cheapest ATA adapters, FXS to SIP available. It's made my Grandstream whose products are notorious for poor quality but they do work and if you're just messing it's the cheapest route other than OTS parts from an electronics supplier. |
| 01:15.46 | avrPhreaque | I am looking for OTS parts to consider, yes |
| 01:16.17 | avrPhreaque | I know little about telephony, specifically, but am an experience electonics designer |
| 01:16.22 | jaytee | plenty of electronic hobbyist suppliers online, You-Do-It in Needham, Massachusetts for one |
| 01:16.53 | avrPhreaque | no, I am looking into commercial parts, like an IC (or group of ICs) to develop the interface with |
| 01:17.05 | avrPhreaque | not interested in hobbyist level crap |
| 01:17.32 | jaytee | but since you're not at "liberty" to discuss details of this mysterious device you wish to build it kinda makes it impossible for us to point you in the proper direction, capiche? |
| 01:17.58 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 01:18.42 | jaytee | avrPhreaque, You-Do-It isn't really "hobby" level per se. Many electronic engineers from Raytheon and other large military sub-contractors in the RT 128 belt or the East Coast Silicon Valley in Massachusetts shop there. |
| 01:19.36 | avrPhreaque | Jaytee: I believe the information I have provided is more than adequate to assit me in locating the information I am seeking. I have a 32 bit processor based PCB, which I want to be able to plug a PSTN telephone instrument into, and have the board provide all services a CO would provide (48VDC, ring current, dial tone, touchtone decoding) |
| 01:20.15 | [TK]D-Fender | avrPhreaque: Go to the LIBRARY, there are tons of books with all this info. |
| 01:20.25 | jaytee | avrPhreaque, check this one out: http://tinyurl.com/c9s634 |
| 01:24.13 | avrPhreaque | TK: maybe in YOUR library...our Library doesn't even carry Popular Electronics |
| 01:24.25 | [TK]D-Fender | avrPhreaque: Imminently Google-able. |
| 01:24.53 | avrPhreaque | jaytee: wonderful...a supplier...now, what, exactly, am I looking for? (I don't see a search bar on that webn site) |
| 01:25.21 | [TK]D-Fender | avrPhreaque: Failing to find such a simple this shows a more serious issue |
| 01:25.30 | octothorp | Couldn't get the inline calls to channel info to show up in the logs, but here it is with manual calls: http://nerdbox.net/pickup.log |
| 01:25.33 | avrPhreaque | fine. Obviously [TK]D-Fender has no interest in helping people. Is it necessary for him to be a comp;lete asshole about ikt, however? |
| 01:25.38 | jaytee | avrPhreaque, call them and ask. |
| 01:25.57 | jaytee | [TK]D-Fender, look! you made another fan! :-) |
| 01:26.05 | avrPhreaque | Jaytee: at 20:25 on a Sunday evening, you betcha!!!! |
| 01:26.40 | jaytee | avrPhreaque, well if it's that urgent, might want to google for Chinese suppliers. I think it's Monday during the day there, they're probably up. |
| 01:26.46 | avrPhreaque | I am not even sure of the name of what i am looking for? SLIC? |
| 01:27.10 | avrPhreaque | NO, it isn;t that urgent, I was just hoping to conduct my search this evening |
| 01:27.20 | jaytee | no one is even sure of what you're looking for |
| 01:27.42 | [TK]D-Fender | avrPhreaque: http://communication.howstuffworks.com/telephone.htm |
| 01:27.44 | jaytee | but they have IC's |
| 01:27.47 | avrPhreaque | how much more specific is it necessary to be? |
| 01:28.13 | avrPhreaque | thanks, I know hwo telephones WORK. |
| 01:28.40 | avrPhreaque | I am looking for references to specif ICs that migth provide the hardware featurs I am looking for, specifixally, to appear to an telephone instrument as the central office |
| 01:28.46 | [TK]D-Fender | avrPhreaque: Yes and it gives specifics on the voltages, etc |
| 01:28.59 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:29.17 | avrPhreaque | I am aware of the Voltages necessary: 48VDC line, 90 VAC - 25 Hz rining current |
| 01:29.20 | [TK]D-Fender | avrPhreaque: Provide tone + voltage. Thats it |
| 01:29.29 | [TK]D-Fender | avrPhreaque: Phones are the dumbest devices in existance |
| 01:30.01 | avrPhreaque | but I need to find a circuit that will also accept the DTMF signal and convert theminto digital numeric values |
| 01:30.14 | jaytee | avrPhreaque, which means you want something that will generate tones like dialtone, busy tone etc as well as supply 48vdc and 90ac ringing. If you're looking for a single chip for that be sure to let us know when you catch the wild goose. |
| 01:30.38 | jaytee | wow, and a tone detector too? |
| 01:30.40 | avrPhreaque | if there os no single chip, where do I find information on what knds of chips I need? |
| 01:30.51 | jql | why not? heh |
| 01:31.03 | octothorp | [TK]D-Fender: Couldn't get the inline calls to channel info to show up in the logs, but here it is with manual calls: http://nerdbox.net/pickup.log |
| 01:31.10 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 01:31.16 | [TK]D-Fender | avrPhreaque: http://www.siongboon.com/projects/2005-07-18_dtmf_circuits/ |
| 01:31.19 | jql | avrPhreaque: buy an FXO card and read off the chip numbers? |
| 01:31.25 | avrPhreaque | jaytee: I keep telling you, I need to offer CO services....that would necessitate interpreting DTMF sequences, yes |
| 01:31.28 | jql | shrugs unhelpfully |
| 01:31.41 | avrPhreaque | what is an FZXO card, and what does it plig in to? |
| 01:31.48 | avrPhreaque | FXO |
| 01:31.56 | jql | google fxo |
| 01:32.01 | [TK]D-Fender | ~fxo |
| 01:32.02 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 01:32.04 | [TK]D-Fender | ~fxs |
| 01:32.05 | jbot | somebody said fxs was foreign exchange station - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 01:32.25 | jaytee | ~101 |
| 01:32.26 | jbot | rumour has it, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 01:32.31 | avrPhreaque | now that URL was helpful, thank you, [TK]D-Fender |
| 01:32.34 | jql | very informative |
| 01:32.55 | avrPhreaque | jbot: that is the opposite of what I am looking for |
| 01:32.56 | jbot | I think you lost me on that one, avrPhreaque |
| 01:33.01 | [TK]D-Fender | avrPhreaque: Yeah, guess where I found it ... http://www.google.ca/search?hl=en&q=DTMF+decoder+circuit&btnG=Google+Search&meta= |
| 01:33.50 | [TK]D-Fender | avrPhreaque: 3rd link, first actual click. |
| 01:33.58 | [TK]D-Fender | avrPhreaque: Only took me about 5 seconds to find |
| 01:34.47 | jaytee | wow! this google thingy is cool! when did they come out with this? |
| 01:36.36 | avrPhreaque | [TK]D-Fender: yes, but that doesn;t answer my question. It only tells me how to decode DTMF, which, while useful, is onely part of my question |
| 01:37.15 | [TK]D-Fender | avrPhreaque: Yes and each individual part of what you want is equally Google-able. this isn't ##electronics |
| 01:38.01 | [TK]D-Fender | avrPhreaque: And from the sound of things the only people in here likely to have the knowledge you are looking for are your COMPETITORS and I doubt they'd be very inclined to help. |
| 01:38.03 | avrPhreaque | so, I am gathering that I will actually require about 4 separate circuits...DTMF decoder, dial/busy tone generator, ring current generator, and an audio intfeface |
| 01:38.05 | avrPhreaque | , yes? |
| 01:38.49 | [TK]D-Fender | avrPhreaque: So far sounds like a "phone -> PC" type deal,.... of which plenty exist |
| 01:38.59 | avrPhreaque | I had thought I had seen, in Electronic Products, over the years, ads for ICs that did it all in one. I was looking for references to that, apparently I misremembered what i had seen |
| 01:39.21 | avrPhreaque | well, sort of, except no PC...dedicated 32 bit embedded processor |
| 01:39.49 | avrPhreaque | I cannot use PCI carss, for example |
| 01:40.26 | avrPhreaque | cards |
| 01:40.43 | avrPhreaque | no standard plugins, I am looking to lay out my own circuit |
| 01:42.08 | avrPhreaque | Well, at least I have more useful information than I had before...thank you, I suppose |
| 01:43.07 | *** part/#asterisk avrPhreaque (i=Default@dsl-dynamic-209-50-30-164.inebraska.com) |
| 01:45.26 | [TK]D-Fender | Octothorp: so far you seem to eb using a bristuff version of that app... |
| 01:45.43 | octothorp | I'm running the hardy packages of asterisk |
| 01:46.42 | octothorp | I was somewhat suprised to see the app available, since it's a bristuff app |
| 01:49.30 | octothorp | Is this close to the way that PickupChan is supposed to work, or am I missing something about it? |
| 01:50.46 | [TK]D-Fender | Octothorp: Looks kinda like it, but I'm not sure on the fine points on that |
| 01:51.01 | [TK]D-Fender | Octothorp: Just looking at the 1.6 native code now.... |
| 01:51.36 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 01:51.45 | [TK]D-Fender | Octothorp: I might suggest playing around with another device to try to see what grabs it. try using a FINXED call toa partial channel name ans see what happens |
| 01:51.55 | [TK]D-Fender | You can do that "in place" |
| 01:52.53 | octothorp | sorry, what do you mean by "FINXED call to a partial channel name"? |
| 01:53.16 | [TK]D-Fender | fixed. As in no the full channel name to pick up |
| 01:53.25 | octothorp | I tried to use cut to pickup "SIP/gizmo" but that had the same result |
| 01:54.04 | *** join/#asterisk chandoo (n=chandra@ool-4353b978.dyn.optonline.net) |
| 01:54.30 | octothorp | I also tried Pickup() and Steal() with similar results |
| 01:55.16 | octothorp | app_intercept and app_valetpark seem like they also might satisfy my need, but I was trying to stay with "stock" apps |
| 01:55.38 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 01:57.41 | [TK]D-Fender | Octothorp : Executing [s@macro-handle_callbacks:203] PickupChan("SIP/gizmo-0822d670", "SIP/spa941-08174590") in new stack |
| 01:57.51 | [TK]D-Fender | Octothorp : No, Gizmo is trying to pick up your SPA |
| 01:58.05 | octothorp | that's what I want it to do |
| 01:58.09 | [TK]D-Fender | Octothorp : the SPA channel is the onle left "ringing" |
| 01:58.40 | [TK]D-Fender | Octothorp: You should try PickupChan(SIP/spa941) first |
| 01:58.59 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 01:59.37 | octothorp | I'm sorry, I mis-spoke, I used cut to try PickupChan(SIP/spa941) |
| 02:01.40 | octothorp | tried it again: -- Executing [s@macro-handle_callbacks:203] PickupChan("SIP/17470141686-08172aa0", "SIP/spa941") in new stack // -- No channel found SIP/spa941. |
| 02:02.11 | octothorp | oops, was hiding my #'s ;-) |
| 02:02.46 | *** join/#asterisk ingenius (n=alektro@host131.190-31-44.telecom.net.ar) |
| 02:03.12 | [TK]D-Fender | OctothorpNot sure what to tell you for this... |
| 02:03.36 | octothorp | it helps to know that I'm not doing something obviously stupid wrong :-) |
| 02:04.19 | octothorp | I guess I'll try another approach. Thanks for your help |
| 02:05.59 | *** join/#asterisk awannabe (n=brad@ip70-162-201-97.ph.ph.cox.net) |
| 02:07.12 | awannabe | hey guys. im trying to concat two variables in the dialplan and use them in-between context. I have did Set(${CHANNEL}_${BLA}=${BLA1}) and when I do a NoOp() after it, it never shows set. that ${BLA1} is being set from a AGI script and when I NoOp() that it has the correct contents |
| 02:09.14 | octothorp | awannabe: maybe you want ImportVar() ? |
| 02:10.12 | [TK]D-Fender | awannabe: Pastebin your failed attempt and related code |
| 02:11.52 | awannabe | [TK]D-Fender, http://pastebin.com/m6bc2f335 is the pastebin |
| 02:12.14 | awannabe | i need to make sure that var is unqiue, so i figured using the channel name is safe, and that LeadID is a value from a AGI script |
| 02:12.35 | [TK]D-Fender | awannabe: exten => _XXXXXXX,14,NoOp(${CHANNELLeadID}) <- not sane |
| 02:12.52 | [TK]D-Fender | awannabe: missing } |
| 02:13.08 | awannabe | missing } ? |
| 02:13.11 | [TK]D-Fender | awannabe: or rather youput it at the end |
| 02:13.47 | [TK]D-Fender | awannabe: ${CHANNEL}LeadID != ${CHANNELLeadID} |
| 02:14.22 | awannabe | [TK]D-Fender, well now that i look at it shoudl 13 be Set(${{CHANNEL}LeadID}=${LeadID}) or something like that? |
| 02:14.36 | awannabe | i see what ya mean now though |
| 02:15.34 | Kobaz | i'm having some weird sip/rtp problems... http://pastebin.com/m3b922470 |
| 02:16.17 | [TK]D-Fender | awannabe: Go stare at it for a bit. If you want help show your failed attempt as well |
| 02:20.16 | *** join/#asterisk siera08 (n=chatzill@218.207.141.90) |
| 02:20.38 | awannabe | [TK]D-Fender, your saing just do ${CHANNEL}${LeadID} basically...yeah that works it seems |
| 02:21.23 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 02:21.26 | Kobaz | [TK]D-Fender: you're the expert on all this stuff... what do you think? |
| 02:21.42 | *** join/#asterisk werdan7 (i=werdan7@freenode/staff/wikimedia.werdan7) |
| 02:22.01 | [TK]D-Fender | Kobaz: Configs please... |
| 02:22.17 | Kobaz | like sip config? |
| 02:22.32 | [TK]D-Fender | Kobaz: Yes |
| 02:22.39 | Kobaz | aye |
| 02:23.54 | Kobaz | http://pastebin.com/m4cafad32 |
| 02:24.50 | Kobaz | i use a db for ast config, so i can't just paste the config file... although... i could format the query results and just that peer |
| 02:24.52 | [TK]D-Fender | Kobaz: No idea |
| 02:25.21 | Kobaz | i can make outgoing calls over that peer just fine |
| 02:25.25 | Kobaz | and incomming works fine |
| 02:25.36 | siera08 | I detected the talking status using NVFaxDetect. but somethimes when the telephone provider plays moh before calling party doesn't accept the call, it recognize moh to a talking status |
| 02:26.04 | siera08 | anyone tell me the way to detect talking status correctly? |
| 02:26.44 | Kobaz | lemme see something... i wonder if if i take the incomming call on voicepulse-primary, and send the other call over voicepulse-backup |
| 02:27.15 | RypPn | ~book |
| 02:27.16 | jbot | [~book] 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://astbook.asteriskdocs.org or see ~buybook |
| 02:29.16 | Kobaz | nope, that didnt work |
| 02:29.31 | Kobaz | it's just so weird |
| 02:32.07 | Kobaz | [TK]D-Fender: if i add an Answer() before the Dial(), it passes audio |
| 02:32.45 | [TK]D-Fender | Kobaz: I did see you supposedly disabled re-invites.... so having * answer shouldn't be an issue |
| 02:32.46 | Kobaz | [TK]D-Fender: but then that will break my findmefollowme handling, a call shouldn't be answered unless it's actually picked up by a human or voicemail, tc |
| 02:33.11 | Kobaz | tc=etc |
| 02:33.26 | Kobaz | supposedly? |
| 02:33.27 | Kobaz | heh |
| 02:33.38 | Kobaz | anyway... is this a bug? |
| 02:33.53 | Kobaz | voicepulse answering the call, should tell asterisk to answer as well, and pass audio |
| 02:34.35 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 02:34.45 | Kobaz | i wonder if voicepulse will allow me to do a reinvite |
| 02:34.47 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 02:40.59 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 02:47.39 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 03:01.09 | *** join/#asterisk xanderp (n=pphillip@c-98-220-163-8.hsd1.in.comcast.net) |
| 03:02.21 | xanderp | Help please: Trying to find a command line way of determining which version of PIAF is running on a system. |
| 03:03.31 | drmessano | Try their forums |
| 03:04.09 | xanderp | thanks... will do some more searching... (was hoping it was a simple command I was just missing) |
| 03:06.25 | drmessano | Could be, but this isnt the best place to ask |
| 03:13.34 | [TK]D-Fender | waves his hand |
| 03:13.41 | [TK]D-Fender | This isn't the channel you are looking for... |
| 03:32.39 | *** join/#asterisk hevnsnt (n=hevnsnt@c-69-254-235-116.hsd1.ks.comcast.net) |
| 03:33.50 | hevnsnt | hello anyone alive in here |
| 03:36.32 | *** join/#asterisk fiddur (n=fiddur@92-32-123-166.tn.glocalnet.net) |
| 03:36.43 | hevnsnt | hey fiddur |
| 03:37.09 | fiddur | Good morning :) |
| 03:38.49 | *** join/#asterisk BBHoss (n=bbhoss@72.146.26.249) |
| 03:39.11 | hevnsnt | think you guys can help out a newbie |
| 03:39.44 | fiddur | hevnsnt: pose your questions... and we'll see |
| 03:40.00 | desdesdesdes | hey |
| 03:40.14 | hevnsnt | trying out my first asterisk setup: Call goes through successfully, (to extension 1000) but no sounds. Get error |
| 03:40.15 | hevnsnt | [Feb 6 17:32:00] WARNING[1828]: file.c:945 ast_streamfile: Unable to open demo-thanks (format 0x4 (ulaw)): No such file or directory |
| 03:40.16 | hevnsnt | [Feb 6 17:32:00] WARNING[1828]: app_playback.c:439 playback_exec: ast_streamfile failed on SIP/192.168.1.5-00597728 for demo-thanks |
| 03:41.05 | fiddur | hevnsnt: Seems it can't find the sounds... for me, the sounds are installed in /var/lib/asterisk/sounds... How did you install it? |
| 03:41.07 | hevnsnt | I have checked and /usr/lib/asterisk/sounds/demo-thanks.gsm exists |
| 03:41.30 | hevnsnt | where is the sound dir configured? |
| 03:42.10 | hevnsnt | well I see my problem.. You said /var/lib/.. |
| 03:42.20 | hevnsnt | Mine installed to /usr/lib/ |
| 03:42.47 | fiddur | hevnsnt: check your asterisk.conf for astvarlibdir I think... |
| 03:43.03 | hevnsnt | ok |
| 03:43.06 | fiddur | I can't see any specific conf for the sounds, but for me they're in the varlib-dir... |
| 03:43.38 | hevnsnt | astvarlibdir => /usr/lib/asterisk |
| 03:44.13 | [TK]D-Fender | hevnsnt: pastebin "core show modules" |
| 03:44.16 | [TK]D-Fender | ~pb |
| 03:44.16 | 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/ |
| 03:44.32 | [TK]D-Fender | hevnsnt: And include your modules.conf |
| 03:44.58 | hevnsnt | thanks working on it |
| 03:45.12 | *** join/#asterisk orlaqp (n=orlatemp@c-98-203-47-18.hsd1.fl.comcast.net) |
| 03:45.44 | orlaqp | hello ways I'm new to asterisk and I'm having problems with X Lite |
| 03:45.56 | orlaqp | hello |
| 03:46.35 | orlaqp | I'm able to dial from the asterisk console but now from the X lite even when it says that I'm connected |
| 03:47.18 | orlaqp | if I dial from the console everything si ok, if I do it from the X Lite in the asterisk console says that it could not find the extension |
| 03:48.54 | [TK]D-Fender | orlaqp: Then whatever you are dialing is not being found in the context your call is using. |
| 03:54.48 | hevnsnt | [TK]D-Fender: I dont have a core show modules |
| 03:55.28 | hevnsnt | here is my modules.conf: http://pastebin.com/m1b3c8f25 |
| 03:55.38 | orlaqp | [TK]D-Fender That could be, let me check the contexts then .. I will get back to you .. thanks |
| 03:55.41 | *** join/#asterisk mandd (n=moo@67.204.24.118) |
| 03:57.45 | orlaqp | [TK]D-Fender Man that was exactly the problem I fix it already .. THANKSS!!!! |
| 03:58.19 | [TK]D-Fender | orlaqp: You're welcome |
| 03:58.34 | [TK]D-Fender | hevnsnt: Taht is a psycho set of load restrictions |
| 03:58.41 | [TK]D-Fender | hevnsnt: try without the "core" |
| 03:58.56 | hevnsnt | got it |
| 03:59.45 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 03:59.59 | hevnsnt | [TK]D-Fender: http://pastebin.com/m26beb3c1 |
| 04:00.01 | hevnsnt | thanks |
| 04:00.59 | [TK]D-Fender | hevnsnt: Why have you cut back all those modules? |
| 04:01.34 | hevnsnt | I have just installed it -- I am running on a OpenWRT install. I am willing to enable whatever |
| 04:02.14 | hevnsnt | I take it I probably need format_wav_gsm.so to play .gsm files |
| 04:02.25 | [TK]D-Fender | hevnsnt: I would re-enable all the codecs if I were you |
| 04:02.30 | [TK]D-Fender | hevnsnt: and formats |
| 04:03.31 | hevnsnt | ok in module.conf do I just remove the ; (eg load => blah) or do I type preload => blah |
| 04:03.55 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 04:04.37 | [TK]D-Fender | hevnsnt: Generally just force a load |
| 04:04.47 | hevnsnt | gottcha thanks for your help |
| 04:04.49 | fiddur | bbl |
| 04:08.24 | hevnsnt | [TK]D-Fender: One last thing, in Modules.conf what is the difference between commented out line "; load => format " and the noload => format... |
| 04:08.58 | [TK]D-Fender | hevnsnt: Did it work? |
| 04:09.08 | hevnsnt | still working on uncommenting all those |
| 04:09.11 | hevnsnt | ;) |
| 04:09.20 | [TK]D-Fender | hevnsnt: Ask me when it works :) |
| 04:09.29 | hevnsnt | fair enough |
| 04:14.16 | hevnsnt | bah |
| 04:18.05 | hevnsnt | still no good http://pastebin.com/m17d6f5e7 |
| 04:18.36 | hevnsnt | looks like my problem is: Unable to find a codec translation path from 0x4 (ulaw) to 0x2 (gsm) |
| 04:20.20 | *** join/#asterisk jmacz (n=jmacz@190.24.103.179) |
| 04:20.47 | *** join/#asterisk tecnico (n=tecnico@75.76.169.148) |
| 04:20.47 | aniasis | ok |
| 04:20.52 | aniasis | i return |
| 04:22.11 | *** join/#asterisk Spirits-Sight (n=christop@c-76-19-220-201.hsd1.ma.comcast.net) |
| 04:23.16 | Spirits-Sight | Could someone tell me the port that I need to forward for asterick correctly I am using SIP / VoicePulse |
| 04:23.36 | [TK]D-Fender | ~sipnat |
| 04:23.37 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 04:23.42 | [TK]D-Fender | Spirits-Sight: Go read ^ |
| 04:23.54 | Spirits-Sight | thanks |
| 04:24.25 | [TK]D-Fender | hevnsnt: if you loaded all the codecs & formats you sholdn't have that issue. After changing that file however you'll ahve to compeltes restart * |
| 04:24.50 | hevnsnt | just an asterisk restart correct |
| 04:24.53 | hevnsnt | I am typing |
| 04:24.56 | hevnsnt | asterisk |
| 04:24.59 | hevnsnt | asterisk -r |
| 04:25.04 | hevnsnt | then stop now |
| 04:25.12 | *** join/#asterisk Vec (n=Vec@196-210-172-92-rrba-esr-3.dynamic.isadsl.co.za) |
| 04:25.23 | hevnsnt | (file changes) then start over |
| 04:26.36 | [TK]D-Fender | hevnsnt: that should do it. "show modules like codec" should show which ones you have loaded (1.4+) |
| 04:27.41 | hevnsnt | interesting |
| 04:27.56 | hevnsnt | codec_ulaw.so shows loaded, but not gsm |
| 04:29.51 | [TK]D-Fender | hevnsnt: make sure you even have it |
| 04:29.58 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 04:30.03 | [TK]D-Fender | hevnsnt: Check your modules folder |
| 04:30.07 | aniasis | [TK]D-Fender, you said you would point me to ITSP and telcos where I can get cheap int'l rates |
| 04:30.28 | [TK]D-Fender | ~itsplist-us |
| 04:30.29 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 04:30.45 | aniasis | ~itsplist-intl |
| 04:32.28 | aniasis | ~itsplist-fr |
| 04:32.37 | aniasis | what about International? |
| 04:32.52 | hevnsnt | [TK]D-Fender: yeah thats obviously the problem.. The gsm file exists, but is not loading |
| 04:33.27 | [TK]D-Fender | aniasis: Not how the list works. they are providers IN the US, not only for calls TO IT |
| 04:33.55 | [TK]D-Fender | hevnsnt: Show me the module. |
| 04:34.13 | hevnsnt | let me poke through asterisk -vvv real quick |
| 04:35.01 | hevnsnt | [Feb 6 18:30:39] WARNING[2082]: loader.c:359 load_dynamic_module: Error loading module 'codec_gsm.so': File not found |
| 04:35.44 | [TK]D-Fender | hevnsnt: Yup, theres the problem |
| 04:36.25 | hevnsnt | root@OpenWrt:/usr/lib/asterisk/modules# ls codec_gsm.so |
| 04:36.25 | hevnsnt | codec_gsm.so |
| 04:39.00 | aniasis | bandwidth.com is not respected ITSP |
| 04:39.01 | aniasis | ? |
| 04:40.32 | drmessano | They're not? |
| 04:40.33 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-0cbd1079f795b0cf) |
| 04:40.56 | hevnsnt | [TK]D-Fender: woot thanks for your help |
| 04:41.21 | hevnsnt | (installed libgsm) |
| 04:41.29 | [TK]D-Fender | hevnsnt: You're welcome |
| 04:44.36 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 04:49.22 | *** join/#asterisk fiddur (i=fiddur@c042.rit.se) |
| 04:52.37 | *** join/#asterisk jmacz (n=jmacz@190.24.103.179) |
| 04:58.23 | *** part/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 05:09.13 | awannabe | [TK]D-Fender, i still cant get that variable to set right. ive tried: Set(${{CHANNEL}LeadID}=${LeadID}), Set(${CHANNEL}${LeadID}=${LeadID}) not sure what else to try |
| 05:09.46 | [TK]D-Fender | awannabe: and I don't see your latest dialplan attempt & failure |
| 05:11.30 | awannabe | [TK]D-Fender, http://pastebin.com/m782190e3 is the dialplan |
| 05:12.27 | [TK]D-Fender | exten => _XXXXXXX,14,NoOp(${CHANNELLeadID}) <--- looks just as broken as before |
| 05:12.43 | [TK]D-Fender | awannabe: that var does not exist |
| 05:13.06 | awannabe | wehn im doing the set, it's not making it one variable? thats what im trying to do |
| 05:13.32 | [TK]D-Fender | awannabe: Your Set is taking the VALUES of those to cars to create the NAME of the new variable |
| 05:13.43 | [TK]D-Fender | vars* |
| 05:14.02 | awannabe | so a new var gets set each time this AGI/call is placed, and have it referenced by one var, as the if i just use ${LeadID} and two people hit the same diaplan extension it will get overwritten |
| 05:14.23 | awannabe | or is their a way to have a variable use the same name but different values for each time a new channel/call is made? |
| 05:14.33 | [TK]D-Fender | exten => _XXXXXXX,13,Set(${{CHANNEL}LeadID}=${LeadID} <- missing brackets, etc. Big mess |
| 05:14.51 | [TK]D-Fender | awannabe: HUH?! |
| 05:15.03 | [TK]D-Fender | awannabe: Variables are unique to each channel... |
| 05:15.25 | awannabe | [TK]D-Fender, so they are unique unless you set them globally, right? |
| 05:15.51 | [TK]D-Fender | awannabe: Yes |
| 05:16.43 | awannabe | well im just a plain iddiot then i guess |
| 05:17.04 | awannabe | yeah i see that, i just tested and it works just fine between context, i must have had something all screwed up before, wow |
| 05:17.37 | *** join/#asterisk _BBV_ (n=buklov@213.138.71.254) |
| 05:18.05 | [TK]D-Fender | awannabe: It is just a giant mess of "OMG I just don't get variables" :) |
| 05:18.38 | awannabe | [TK]D-Fender, no crap! wow i think i thought it was not working before cause the AGI script was broke, so that was a doubel id10t error on my side hah |
| 05:19.38 | [TK]D-Fender | awannabe: Add a few more braces, how much worse can it ge? :) |
| 05:20.54 | awannabe | [TK]D-Fender, true, as long as it doesnt segfault anything were get! |
| 05:21.22 | awannabe | [TK]D-Fender, thanks for help and clarification on that, i thought that's how they all worked, then i guess with my AGI fiasco i second guessed myself heh |
| 05:27.12 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 05:37.36 | [TK]D-Fender | Alrighty... checkout time, later all |
| 05:43.06 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 05:43.23 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 05:51.25 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 06:02.48 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
| 06:12.37 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 06:16.09 | *** join/#asterisk SunnyDP (n=scan@bas7-montrealak-1128544643.dsl.bell.ca) |
| 06:16.46 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 06:24.15 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 06:34.19 | *** join/#asterisk xrmx__ (n=rm@host128-22-dynamic.15-87-r.retail.telecomitalia.it) |
| 06:48.27 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 06:56.05 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 07:02.17 | *** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
| 07:02.38 | brunner | Has this been completely phased out in favor of ARA, or can I still use this for now in 1.4? http://www.voip-info.org/wiki/view/Asterisk+voicemail+database |
| 07:04.02 | pif | hi, how do you guys configure sntp support on polycoms? |
| 07:04.41 | pif | my sntp server's localtime has adjusted to dst yesterday, but the polycoms stay at the old time |
| 07:10.20 | *** join/#asterisk octothorp (n=vandebo@abydos.nerdbox.net) |
| 07:11.22 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) |
| 07:13.39 | octothorp | I finnally got my attempted solution to work (with StealChan), but it doesn't bridge the two channels, it uses ast_channel_masquerade to clobber one of them |
| 07:13.53 | octothorp | Is there something like StealChan that connects the two channels together? |
| 07:16.12 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 07:23.38 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:25.37 | *** join/#asterisk c0rnoTa (n=c0rnoTa@78.24.154.158) |
| 07:25.41 | *** part/#asterisk c0rnoTa (n=c0rnoTa@78.24.154.158) |
| 07:27.51 | *** join/#asterisk shtoom (n=shtoom@125.16.8.110) |
| 07:28.58 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 07:37.25 | *** join/#asterisk dr_gogeta86 (n=fisgro@81-208-88-100.ip.fastwebnet.it) |
| 07:47.04 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 07:51.03 | *** join/#asterisk sapata (n=sapata@78.128.10.18) |
| 07:53.57 | *** join/#asterisk Chris-NB (n=chris@85-126-61-10.work.xdsl-line.inode.at) |
| 07:53.59 | Chris-NB | hi |
| 07:55.00 | Chris-NB | I've a central office and 4 subsidiaries. every office gets an asterisk. each office is connected to each other |
| 07:55.13 | Chris-NB | I want to interconnect the offices via iax |
| 07:56.00 | Chris-NB | what would be the best option? register each office on the other 3? |
| 07:58.49 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 07:59.02 | kimo_sabe | /quit blah |
| 07:59.42 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 07:59.43 | *** mode/#asterisk [+o denon] by ChanServ |
| 08:02.37 | *** join/#asterisk raden_work (n=jon@adsl-99-139-235-165.dsl.applwi.sbcglobal.net) |
| 08:09.30 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 08:30.04 | angryuser | Chris-NB, yes it's the simples way |
| 08:30.09 | angryuser | simplest* |
| 08:30.45 | Chris-NB | angryuser, okay. do you know of an configuration example? |
| 08:31.07 | angryuser | Chris-NB, have you ever configured iax/sip provider ? |
| 08:31.44 | Chris-NB | angryuser, jep. but only for one single provider, not for multible peers |
| 08:33.03 | angryuser | Chris-NB, it's the same stuff, separs each office with local number's for example office 1 = peers 1000 office 2 peers 2000 , ect and aftewards do your dialplan, cal leach server with that internal numbers |
| 08:33.15 | angryuser | separe* |
| 08:33.55 | Chris-NB | okay, I'll try. thanks |
| 08:45.10 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 09:09.16 | *** join/#asterisk shtoom (n=shtoom@125.16.8.110) |
| 09:10.14 | *** join/#asterisk Mw3 (i=mw3@mw3.hu) |
| 09:15.17 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-97efa432e8f41422) |
| 09:16.38 | *** join/#asterisk AD2008 (n=RatX@wbs-41-208-233-241.wbs.co.za) |
| 09:17.45 | phr3ak | i plan to buy an openvox card. how possible "transfer" the incoming pstn call to fax receiver application on a specified extension? anyone could show me an example config line? |
| 09:18.04 | *** join/#asterisk Dovid (n=annon@tony09-118-62.inter.net.il) |
| 09:22.32 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 09:40.42 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 09:46.11 | *** join/#asterisk andrebarbosa (n=andrebar@212.13.49.67) |
| 10:03.26 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:09.07 | *** join/#asterisk Chodorenko (n=chodoren@ext.one.by) |
| 10:09.55 | *** join/#asterisk zaafouri (n=zaafouri@196.203.51.238) |
| 10:14.29 | *** join/#asterisk shtoom (n=shtoom@59.165.83.171) |
| 10:15.51 | *** join/#asterisk Mw3 (i=mw3@mw3.hu) |
| 10:16.56 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 10:29.20 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 10:30.32 | *** join/#asterisk bminish (n=bminish@pdpc/supporter/professional/bminish) |
| 10:46.37 | *** join/#asterisk zaffa (n=x@83-216-20-253.dsl.maxinetti.fi) |
| 10:47.45 | *** join/#asterisk xnik (n=Nikolay@ra18.cdma.bwc.ru) |
| 10:47.45 | *** join/#asterisk beherit (n=albert@203.153.6.16) |
| 10:47.59 | xnik | hi |
| 10:48.08 | xnik | i need help |
| 10:48.31 | zaffa | hi |
| 10:48.41 | andrebarbosa | Anyone using linksys phones on a g729 scenario? |
| 10:48.49 | andrebarbosa | I'm having problems with transfered calls, because linksys phones can't handle two simultaneous g729 calls |
| 10:48.55 | andrebarbosa | they try to re.invite to g711, but asterisk rejects, and the call hangs.. :( |
| 10:50.03 | xnik | Anyone using asterisk with PBX ? |
| 10:51.31 | phr3ak | everybody |
| 10:52.06 | phr3ak | andrebarbosa: this codec is available in asterisk? |
| 10:52.26 | andrebarbosa | yes |
| 10:52.28 | andrebarbosa | I have licenses |
| 10:52.51 | mort_gib | andrebarbosa: Use ulaw for all calls or buy better phones... |
| 10:53.08 | zaffa | anyone had problem with retransmissions with libpri (1.4.9)? I have... telco is sending RR and * is responding by retransmitting CallRelease messages. |
| 10:53.10 | andrebarbosa | eheh mort_gib |
| 10:53.11 | andrebarbosa | :D |
| 10:53.24 | andrebarbosa | i need g729, because most of phones are in the WAN |
| 10:53.31 | andrebarbosa | need to save bandwith |
| 10:53.35 | mort_gib | andrebarbosa: -Buy better phones.... |
| 10:53.42 | andrebarbosa | right |
| 10:54.04 | andrebarbosa | polycom's can handle two g729 calls |
| 10:54.06 | andrebarbosa | (Y) |
| 10:54.18 | andrebarbosa | but I'm trying to make this work with linksys |
| 10:54.44 | mort_gib | andrebarbosa: I used to think Linksys was a great company, like Cisco affiliate etc, but I have had endless crap with ADSL routers Accesspoints etc |
| 10:55.13 | mort_gib | andrebarbosa: Why Linksys?? -Price?? |
| 10:55.25 | andrebarbosa | yes |
| 10:55.32 | andrebarbosa | And they have portuguese menus |
| 10:55.50 | mort_gib | andrebarbosa: Being stingy is not a good inroad into Asterisk and VOIP |
| 10:56.04 | mort_gib | andrebarbosa: Use Snom, I think they DO portugese |
| 10:56.55 | zaffa | about libpri, anyone? |
| 10:57.04 | andrebarbosa | We had lots of probs with Snom Phones |
| 10:57.08 | andrebarbosa | :s |
| 10:57.19 | mort_gib | andrebarbosa: What kind of problems?? |
| 10:57.22 | andrebarbosa | we have a 370 Snom phone with lastest firmware |
| 10:57.30 | andrebarbosa | and the phone crashes a lot |
| 10:57.30 | andrebarbosa | :s |
| 10:57.34 | mort_gib | Nope |
| 10:57.40 | mort_gib | I don't |
| 10:57.48 | mort_gib | s/I/it |
| 10:58.11 | andrebarbosa | you recommend snom? |
| 10:58.20 | mort_gib | Snom had one firmware out lately that was faulty, the 7.3.XX is fine |
| 10:58.28 | mort_gib | Yeah, they are not too bad |
| 10:58.42 | andrebarbosa | I like polys |
| 10:58.58 | *** join/#asterisk Daejeo (n=chatzill@110.9.169.185) |
| 10:58.58 | mort_gib | andrebarbosa: Everybody do, they are harder to configure though |
| 10:59.05 | andrebarbosa | yes |
| 10:59.13 | andrebarbosa | but we have a autoprovisiong system |
| 10:59.19 | andrebarbosa | it works well with polys |
| 10:59.29 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 10:59.31 | mort_gib | andrebarbosa: Which system?? I use vi |
| 11:00.14 | andrebarbosa | we have an appliance that does the work |
| 11:00.49 | andrebarbosa | detects the phone, and you just assign an extension, and the phone reboot's and downloads the new configuration |
| 11:01.36 | mort_gib | Coll, Polycoms are nice |
| 11:01.42 | andrebarbosa | yep |
| 11:01.55 | andrebarbosa | The system works with Snoms, Linksys and Grandstreams |
| 11:02.03 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 11:02.05 | andrebarbosa | and Aastras |
| 11:02.58 | mort_gib | What appliance?? |
| 11:03.23 | andrebarbosa | i can give you the details in private |
| 11:03.41 | mort_gib | Sure |
| 11:14.00 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 11:24.52 | linuxstb | Hi all. Apart from them having a larger LCD and more buttons, are there any other reasons to prefer the more expensive Polycoms (e.g. IP5xx/IP6xx) over the IP320/330 for use with *? |
| 11:26.02 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 11:41.04 | *** join/#asterisk Mw3 (i=mw3@mw3.hu) |
| 11:49.31 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:49.43 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 11:53.04 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 11:53.12 | *** join/#asterisk hi365 (n=hi365@94.159.131.232) |
| 11:55.46 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 12:02.13 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 12:06.38 | Chodorenko | Hi All heto to i can detect length of Variable >5 sumbols? |
| 12:06.50 | Chodorenko | * Howe to |
| 12:11.14 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:14.46 | *** part/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:15.02 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:15.32 | *** join/#asterisk Victor_Yure_ (n=victor@unaffiliated/victoryure/x-837844) |
| 12:15.42 | *** join/#asterisk saftsack (n=saftsack@p57924F5F.dip.t-dialin.net) |
| 12:33.51 | *** join/#asterisk hi365 (n=hi365@94.159.144.204) |
| 12:34.05 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 12:41.01 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:45.32 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 12:48.27 | *** join/#asterisk Edis0n (n=Edis0n@u16-173.dsl.vianetworks.de) |
| 12:51.23 | *** join/#asterisk sasargen (n=chatzill@173-100-50-208.pools.spcsdns.net) |
| 12:53.05 | *** join/#asterisk juanjoc (n=jcomella@200.69.219.113) |
| 12:53.21 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 12:54.39 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 12:54.55 | *** join/#asterisk zchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com) |
| 12:58.03 | *** join/#asterisk no_nick_here (n=bobnance@76.29.229.137) |
| 13:00.13 | no_nick_here | good morning! |
| 13:06.37 | *** join/#asterisk clintc (n=clintc@n128-227-7-200.xlate.ufl.edu) |
| 13:12.26 | Chodorenko | no_nick_here: |
| 13:14.01 | no_nick_here | yes? |
| 13:17.08 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:17.52 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 13:19.35 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 13:23.51 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 13:25.41 | jaytee | morning beek, [TK]D-Fender |
| 13:27.29 | beek | Good morning jaytee |
| 13:28.13 | *** join/#asterisk mort___ (n=mort@user-54405220.wfd76b.dsl.pol.co.uk) |
| 13:28.28 | *** join/#asterisk path_ (n=path_@71-119-21-190.adsl.terra.cl) |
| 13:31.23 | *** join/#asterisk jplank (n=gbove@cpe-075-181-097-208.carolina.res.rr.com) |
| 13:31.38 | *** join/#asterisk RobertLaptop (n=rmiddle@63.68.135.4) |
| 13:33.52 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 13:34.48 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 13:34.48 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:38.22 | *** join/#asterisk jmacz (n=jmacz@190.146.58.46) |
| 13:39.58 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:41.51 | *** join/#asterisk Mw3 (i=mw3@mw3.hu) |
| 13:41.55 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 13:42.43 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 13:50.24 | *** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca) |
| 13:50.52 | timeshell | Happy Monday!! |
| 13:52.11 | *** join/#asterisk Badrobot- (n=Badrobot@cpe-76-173-233-75.socal.res.rr.com) |
| 13:54.15 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 13:54.21 | ttl- | hi |
| 13:55.08 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 14:03.18 | *** join/#asterisk RattX (n=RatX@wbs-41-208-236-146.wbs.co.za) |
| 14:03.19 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 14:04.51 | *** join/#asterisk dvdevel (n=devel@wiggum.digitalcoven.com) |
| 14:11.06 | *** join/#asterisk whist15 (i=whistler@96-35-205-179.dhcp.stls.mo.charter.com) |
| 14:11.15 | whist15 | hi, Im having some trouble with the managers interface on asterisk |
| 14:11.41 | *** join/#asterisk eric_hill (i=eric_hil@204.94.175.11) |
| 14:11.42 | whist15 | I manually coneect with telnet, but keep getting a login error |
| 14:11.58 | whist15 | ik believe my manager.conf is setup correctly |
| 14:12.06 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 14:12.32 | whist15 | is tehre something I am missing? |
| 14:12.50 | eric_hill | I'm seeing the queue_log rotate to a queue_log.1 file, but subsequent entries go into the .1 file, not the main log? |
| 14:14.04 | [TK]D-Fender | whist15: Would help if you showed us the problem. |
| 14:14.05 | *** part/#asterisk AndyML (n=AndyML@pool-96-245-116-32.phlapa.fios.verizon.net) |
| 14:14.07 | [TK]D-Fender | ~pb |
| 14:14.08 | 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/ |
| 14:14.10 | [TK]D-Fender | ^^^^ |
| 14:14.26 | *** join/#asterisk moy (n=chatzill@74.12.124.89) |
| 14:15.28 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 14:16.54 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 14:18.27 | MaliutaLap | failing that it might be fixed at http://dinosaursfuckingrobots.com/ ;) |
| 14:19.57 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 14:25.17 | *** join/#asterisk jerlique (i=d208a802@gateway/web/ajax/mibbit.com/x-3bfb88aebc5ad1bd) |
| 14:28.28 | jerlique | Apparently Asterisk 1.6 has fax support. Can someone point me to the relevant web pages with the docs? |
| 14:29.21 | mmlj4 | http://voip-info.com and search for fax |
| 14:30.08 | jerlique | I'll double check, but I though that was the addon packages, not the inbuilt one |
| 14:30.42 | drmessano | Its T.38 passthru |
| 14:32.26 | jerlique | drmessano: Do you mean app_fax is merely pass throug support? |
| 14:32.38 | jerlique | (in asterisk 1.6.x) |
| 14:32.44 | drmessano | yes |
| 14:32.57 | drmessano | Just as I said |
| 14:33.06 | jerlique | Just checking ;) |
| 14:33.34 | coppice | app_fax isn't passthrough |
| 14:33.49 | mvanbaak | drmessano: that's not true |
| 14:34.01 | drmessano | Its not? |
| 14:34.18 | mvanbaak | drmessano: have a look at: core show application ReceiveFax |
| 14:35.00 | drmessano | Nope |
| 14:35.07 | mvanbaak | unless you define passthru as 'fax comes in and we save the data on disk' |
| 14:35.08 | *** join/#asterisk macros73_ (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 14:35.51 | drmessano | Your application(s) is (are) not registered |
| 14:36.10 | mvanbaak | ah, you did not build asterisk while spandsp was installed |
| 14:36.37 | mvanbaak | <PROTECTED> |
| 14:36.41 | mvanbaak | Receives a fax from the channel into the given filename overwriting |
| 14:36.43 | drmessano | Sure did |
| 14:36.46 | mvanbaak | the file if it already exists. File created will have TIFF format. |
| 14:37.18 | mvanbaak | this is Asterisk SVN-branch-1.6.0-r184763 |
| 14:37.42 | drmessano | Same here |
| 14:38.09 | drmessano | Hmmm |
| 14:38.18 | mvanbaak | run: make menuselect |
| 14:38.30 | jerlique | Is app_fax reliable? How does it compare to iaxmodem+hylafax? |
| 14:38.33 | mvanbaak | apps/app_fax should be checked. if not it will show you why not |
| 14:38.36 | drmessano | app_fax would have been selected |
| 14:39.05 | mvanbaak | yeah, and if not selectable it will list why not on the bottom of your screen. |
| 14:39.22 | drmessano | I know how it works |
| 14:39.26 | drmessano | I know its been selected |
| 14:39.40 | coppice | jerlique: depends on your machine. iaxmodem pretty much always works for rx and tx. with app_fax rx is usually fine, but tx may have problems because asterisk can't get its timing right. |
| 14:39.47 | mvanbaak | module load app_fax.so |
| 14:40.24 | coppice | if Digium are plugging a commercial FAX package in for SwitchVOX, I guess they'll need to get that timing sorted out :-) |
| 14:40.41 | jerlique | coppice: How do you fix the timing? |
| 14:40.43 | drmessano | I will have to dig later.. Thought all it did was passthru... Seems like it was noted that 1.6.0 wouldnt rx or send *yet*, with it coming soon.. I know app_fax has been selected.. Guess I need to see whats broken if thats not the case |
| 14:41.32 | mvanbaak | drmessano: it can do both. no Idea why it's not loaded on your machine. Good luck with the digging |
| 14:41.47 | coppice | jerlique: its an asterisk problem affecting all apps. most apps don't care as much, though. if you send from one * machine to another, and record both ends, you'll see gaps in the received audio |
| 14:42.46 | jerlique | Is that why ChanSpy audio can sometime be "wobbly" |
| 14:44.13 | coppice | strangely, IAX data passing through the same box doesn't see to suffer from the same problem, so iaxmodem is more reliable. some boxes have trouble. others don't. I don't see the problem on my own boxes, but I've had logs sent to me showing it |
| 14:44.46 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:44.49 | jerlique | coppice: Do you use it with Hylafax or something else? |
| 14:44.56 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 14:45.45 | coppice | iaxmodem works OK with hylafax, efax, and even Windows FAX, accessing iaxmodem through virtual terminal servers. I'm not sure how robust that last option is, though |
| 14:47.20 | jerlique | Do you mean Vmware "modems" when you say virtual terminal servers? |
| 14:47.38 | mmlj4 | do what? |
| 14:48.08 | coppice | I'm not sure of the details of how that was set up, but I know people have done it |
| 14:54.17 | jerlique | coppice: thats cool. lack of modem support in vmware esxi sucks. I'm tossing up between hylafax+ and efax any suggestions? |
| 14:54.33 | coppice | hylafax+ |
| 14:55.23 | coppice | hylafax+ and iaxmodem were literally made for each other |
| 14:55.43 | MaliutaLap | they should get married and have kids |
| 14:55.51 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:55.52 | jerlique | lol |
| 14:56.04 | coppice | that's not a nice thing to wish on anyone |
| 14:56.15 | jerlique | then get divorced and take half the faxes with them |
| 14:56.17 | MaliutaLap | they'd have faxlets |
| 14:58.50 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 15:03.07 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 15:03.10 | whist15 | i keep getting an unable to request cahnnel when using a click to dial script in the log |
| 15:03.15 | *** part/#asterisk Edis0n (n=Edis0n@u16-173.dsl.vianetworks.de) |
| 15:03.28 | whist15 | I cant find much on google in regards to the message |
| 15:03.51 | whist15 | whay would be the cause? I can connect to the admin interface, it is setup correctly in sip.conf |
| 15:03.56 | whist15 | is tehre something else Im missing? |
| 15:08.39 | [TK]D-Fender | whist15: AMI != sip.conf |
| 15:08.57 | [TK]D-Fender | whist15: And you aren't showing us anything so we can't help. |
| 15:09.02 | [TK]D-Fender | ~pb |
| 15:09.03 | 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/ |
| 15:09.03 | jermey_g | hi |
| 15:09.04 | [TK]D-Fender | whist15: ^^^^^ |
| 15:09.13 | jermey_g | hi |
| 15:09.30 | jermey_g | where can i downloaded asterisk 1.2 last stable release |
| 15:10.13 | [TK]D-Fender | jermey_g: www.asterisk.org |
| 15:12.35 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 15:12.36 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:12.52 | mort_gib | Scaling question, I need to stick 120 users on asterisk, with about 45 concurrent calls with on T1/E1 Can I fi them on one server?? |
| 15:13.01 | *** join/#asterisk mweichert (n=mweicher@216.13.154.21) |
| 15:13.02 | mort_gib | s/fi/fit |
| 15:13.16 | [TK]D-Fender | mort_gib: You can do far more |
| 15:13.35 | mweichert | Hello #asterisk :) |
| 15:13.39 | mort_gib | [TK]D-Fender: So I don't need to look at SER |
| 15:13.41 | [TK]D-Fender | mort_You know Digium sells 4 port E1 cards right? Thats 120 channels :) |
| 15:13.44 | MaliutaLap | [TK]D-Fender: and if said server is a PIII700? |
| 15:13.46 | *** join/#asterisk [1]hi365 (n=hi365@94.159.144.204) |
| 15:13.59 | [TK]D-Fender | mort_gib: of just termination, so duh, you can clearly scale twice as large :) |
| 15:14.01 | [1]hi365 | anyone using an audiocodes mp203? |
| 15:14.29 | mort_gib | No, server is a Dell 2970, with Dual Quadocore CPU and 8GB ram |
| 15:14.34 | mweichert | I'm trying to diagnose some dropped calls. Calls are dropped as soon as they are picked up. Doing some sniffing, I see that my phone is sending a BYE request for some reason just after a few seconds that I pick up the call... how can I determine why this is occuring? |
| 15:14.50 | [TK]D-Fender | mort_gib: You could get away with a 1st gen P4 :) |
| 15:14.56 | [TK]D-Fender | mort_gib: and 256 meg :) |
| 15:15.07 | [TK]D-Fender | mort_gib: I would personally want at least 512, but here... |
| 15:15.09 | MaliutaLap | mort_gib: I was pointing out that "server" could be anything and you can't make assertions without information |
| 15:15.29 | [TK]D-Fender | mort_gib: Your server is overkill, have fun |
| 15:15.30 | mort_gib | Thanks guys! :-) |
| 15:15.40 | MaliutaLap | mort_gib: my * server is a Coppermine 700 (PIII gen celeron) with 400MB of ram |
| 15:16.02 | mort_gib | Well this is a rather big install, chances are the client will go for two servers and a media gateway |
| 15:16.27 | jermey_g | [TK]D-Fender:where in asterisk.org |
| 15:16.36 | [TK]D-Fender | jermey_g: Downloads |
| 15:16.45 | [TK]D-Fender | jermey_g: Show some initiative and LOOK |
| 15:16.56 | mort_gib | I run 8 Asterisk servers, smallest is a Soekris 5501 that supports 8 users fine |
| 15:17.48 | Chodorenko | jermey_g: http://downloads.digium.com/pub/asterisk/releases/ |
| 15:18.30 | Chodorenko | jermey_g: http://downloads.digium.com/pub/asterisk/releases/asterisk-1.2.31.1.tar.gz |
| 15:18.58 | Chodorenko | jermey_g: http://downloads.digium.com/pub/asterisk/releases/asterisk-addons-1.2.9.tar.gz |
| 15:19.35 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:20.08 | jermey_g | [TK]D-Fender:#@#@$&*($#&*(@ |
| 15:20.28 | jermey_g | Thanks Chodorenko, i had to copy link location of the 1.4 release to get that |
| 15:21.00 | jermey_g | mort_gib:what kind of hardware do you run * on |
| 15:22.20 | mort_gib | For small setups Soekris 5501's bigger setups Dell servers.... |
| 15:24.04 | MaliutaLap | I don't like Dell kit |
| 15:24.13 | MaliutaLap | prefer HP ILO to DRAC |
| 15:25.40 | mort_gib | MaliutaLap: I have more problems with users to be honest :-) |
| 15:25.56 | MaliutaLap | mort_gib: I don't talk to users anymore |
| 15:26.04 | MaliutaLap | mort_gib: users are a concept |
| 15:26.15 | jermey_g | mort_gib:have you tried it in on HP Proliant quad code |
| 15:26.17 | jermey_g | core |
| 15:26.21 | coppice | well, users were conceived |
| 15:26.23 | MaliutaLap | thank f@#king god |
| 15:26.32 | mort_gib | MaliutaLap: Yes, well... users hmm No, don't like HP stuff much |
| 15:27.05 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:27.23 | mort_gib | The local HP pushers are really arrogant and expensive |
| 15:27.38 | MaliutaLap | I find peopl |
| 15:27.43 | MaliutaLap | suppliers that suit me |
| 15:28.04 | MaliutaLap | I have opportunities though |
| 15:28.06 | mort_gib | MaliutaLap: Which you can do in a country that supports more than one dealer.... |
| 15:28.28 | MaliutaLap | mort_gib: what country you in? |
| 15:28.41 | jermey_g | with 1.2 which asterisk addons should i use 1.x? |
| 15:28.42 | mort_gib | MaliutaLap: This works is for a client in Gibraltar |
| 15:29.07 | *** join/#asterisk axisys (n=axisys@117.18.228.248) |
| 15:29.10 | MaliutaLap | mort_gib: you can't use a british supplier who has offices there? |
| 15:29.13 | [TK]D-Fender | jermey_g: 1.2.x <- |
| 15:29.57 | mort_gib | MaliutaLap: No, there is only one local HP pusher, not counting the Spanish guys supporting the betting companies |
| 15:31.13 | *** join/#asterisk jjshoe (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 15:31.17 | MaliutaLap | I knew Gibraltar was small, but I figured it to be about the same size as Brisbane |
| 15:31.29 | MaliutaLap | and I have more than one option here |
| 15:31.58 | mweichert | what are some causes for a user agent to send SIP/Bye ? |
| 15:32.29 | mweichert | or how can you identify at the transport/network level why a SIP/Bye request was sent? |
| 15:32.33 | MaliutaLap | mweichert: hang up, they don't like you ... |
| 15:32.46 | [TK]D-Fender | mweichert: Look at the SIP debug for the call |
| 15:33.21 | MaliutaLap | transport/network level??? any info is going to be higher up the stack than that |
| 15:34.04 | whist15 | what would be the resons behind not being able to request a sip channel? |
| 15:34.13 | [TK]D-Fender | Where I come from SIP = Application Layer..... |
| 15:34.27 | stevetotaro | wow, stock market is doing great |
| 15:34.35 | mweichert | [TK]D-Fender, on the user agent client? |
| 15:34.37 | [TK]D-Fender | whist15: Not having a host to contact, etc |
| 15:34.44 | [TK]D-Fender | mweichert: on * <- |
| 15:34.45 | whist15 | Im trying to setup a click to dial script, i can connect tothe admin interface, I can login succesfully, but I cant dial? |
| 15:34.51 | mort_gib | MaliutaLap: Gibraltar is like 35000 inhabitants, and a lot of border workers |
| 15:35.20 | stevetotaro | border workers? |
| 15:35.21 | [TK]D-Fender | whist15: Wake up and PASTEBIN your setup and debug of your failed attempts <- |
| 15:35.23 | [TK]D-Fender | ~pb |
| 15:35.24 | 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/ |
| 15:35.26 | [TK]D-Fender | ^^^^^^ |
| 15:35.32 | mweichert | [TK]D-Fender, I don't have access to the Asterisk server |
| 15:35.43 | mort_gib | stevetotaro: People living is Spain and working in Gibraltar, like I do |
| 15:36.09 | stevetotaro | so you go home at the end of the day? |
| 15:36.12 | [TK]D-Fender | mweichert: "Help me fix my car", "Ok, lets pop the hood open", "but I can't!!!!" , "then why'd you bother asking?!?!" |
| 15:36.19 | mort_gib | stevetotaro: Don't we all :-) |
| 15:36.23 | stevetotaro | not like the US's mexicans |
| 15:36.35 | mort_gib | stevetotaro: No, we get paid better ;-) |
| 15:37.52 | mweichert | [TK]D-Fender, fair enough. I'll try to call my provider and see if I can get them to get me a log. Thanks for your help. |
| 15:38.31 | stevetotaro | the mexicans stay here |
| 15:39.13 | mort_gib | stevetotaro: Gibraltar is too damn small, all development is, and has been on reclaimed land for years |
| 15:41.22 | *** join/#asterisk jstew (n=jstewart@fw-ext.fusionary.com) |
| 15:41.31 | jstew | Greetings. |
| 15:41.55 | mweichert | [TK]D-Fender, just to be sure - even though the BYE request is sent from the phone, the problem could still be on the Asterisk server? |
| 15:42.08 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 15:42.30 | [TK]D-Fender | mweichert: Mybe the phone said "BYE" because it didn't like something * said |
| 15:42.41 | jstew | I'm writing an application that uses TxFax and it gives no feedback via AMI events. Is it possible to get the return status of txfax in my dialplan then send an AMI event from there? |
| 15:42.58 | mweichert | [TK]D-Fender, ok. Thanks. Hopefully I can get the service provider to co-operate. |
| 15:43.17 | stevetotaro | http://speakbrain.smfforfree3.com/index.php?topic=151.0 |
| 15:46.08 | mort_gib | Is the Polycom IP 320 a good "normal desk phne" ?? |
| 15:46.41 | Qwell | mort_gib: I've not used a 320, but my 430 is very nice |
| 15:46.50 | jstew | My 550 is wonderful too. |
| 15:47.01 | stevetotaro | 320 is ok |
| 15:47.08 | jstew | Provisioning is a nightmare though unless you really love XML |
| 15:47.23 | stevetotaro | or web interface |
| 15:47.33 | [TK]D-Fender | Far from a nightmare, but definitely a steeper learning curve/ |
| 15:47.36 | stevetotaro | for a couple phones, i just use the web interface |
| 15:47.57 | mort_gib | jstew: I don't have problems with XML, I looked into the IP 650 and after a bit of frustration it worked |
| 15:48.09 | [TK]D-Fender | People configuring Polycom's via the web interface should be dragged out and shot. Survivors should be shot AGAIN. |
| 15:48.10 | stevetotaro | the web server comes up much quicker now, it used to take like five minutes after a reboot |
| 15:48.16 | mort_gib | stevetotaro: This is 150+ lusers |
| 15:48.32 | stevetotaro | yeah, XML is the way to go |
| 15:48.33 | BrianR___ | you really only need to change a few things from the default config settings, so the polycom xml thing isn't as bad as it seems at firs |
| 15:48.33 | BrianR___ | t |
| 15:48.42 | mort_gib | [TK]D-Fender: Yes, well if you have ONE single Polycom.... |
| 15:48.44 | BrianR___ | Just the dialplan and the sip auth really... |
| 15:48.51 | jstew | the web interface makes me want to cry though. |
| 15:48.54 | *** join/#asterisk jstoker (n=jstoker@88-107-192-59.dynamic.dsl.as9105.com) |
| 15:48.55 | KavanS | does anyone here do video w/asterisk? |
| 15:48.58 | [TK]D-Fender | loads a round into the chamber... |
| 15:49.09 | stevetotaro | inserts 48 round mag |
| 15:49.19 | stevetotaro | one already in chamber |
| 15:49.24 | mort_gib | jstew: Snoms comes with nice a webinterface |
| 15:49.36 | *** join/#asterisk workdraft (n=ian@122.52.117.21) |
| 15:49.40 | mmlj4 | and not much else nice, IMHO |
| 15:49.45 | BrianR___ | Not all of the configurables are in the web interface anyway. I bet it has less than 10% coverage. |
| 15:49.58 | mort_gib | So the IP 320 is a nice "normal mortal phone" ?? |
| 15:50.11 | BrianR___ | I used 430's at my last job... |
| 15:50.16 | stevetotaro | i really wish it had the FTP/TFTP settings in the freaking web interface!!!!!!111111 |
| 15:50.21 | [TK]D-Fender | mort_gib: your typical low-volume Cubicle user |
| 15:50.24 | mmlj4 | unless they've changed in the last few years, anyhow |
| 15:50.27 | jstew | Most web interfaces suck on IP phones. |
| 15:50.38 | BrianR___ | Only gripe is that the 430 had an easily accesible speakerphone which sucked for the neighborhood when cube dwellers figured out how to use it, |
| 15:50.49 | workdraft | got my asteriskNow work with bandwidth.com. i am now able to call an outside line but i can't hear their voice from my end. but they can hear me. any piece of advice? |
| 15:50.55 | stevetotaro | but if i could tell it via the web interface to get configs from such and such FTP then i would be happy |
| 15:51.01 | mort_gib | mmlj4: I really DO have more issues with users, like operating a phone with a flat hand! |
| 15:51.06 | [TK]D-Fender | BrianR___: Shame on them for making their product easy to use! |
| 15:51.11 | mmlj4 | workdraft: are you behind NAT? |
| 15:51.22 | workdraft | my ip phone, yes |
| 15:51.26 | BrianR___ | [TK]D-Fender: I never did find out how to administratively disable the speakerphone... |
| 15:51.39 | mmlj4 | mort_gib: how's that? I'm trying to find something nice to say about snom |
| 15:51.43 | BrianR___ | [TK]D-Fender: I suppose I could have fucked with the gain settings on it and made it unusable, but... |
| 15:51.43 | [TK]D-Fender | BrianR___: Not sure that you can... |
| 15:51.45 | stevetotaro | step 1, open phone |
| 15:51.53 | stevetotaro | step 2, cut speaker wires |
| 15:51.56 | mort_gib | [TK]D-Fender: Well this is stupid, they want a price, but they don't know what phones they want.... |
| 15:52.00 | BrianR___ | [TK]D-Fender: so glad I escaped that stuff ) |
| 15:52.01 | stevetotaro | step 3. put phone back together |
| 15:52.03 | workdraft | @mmlj4: and my asterisk server also. |
| 15:52.04 | mmlj4 | I've done that once |
| 15:52.10 | [TK]D-Fender | BrianR___: that's an ugly hack I know you can already see coming to bite you in the ass :) |
| 15:52.14 | *** join/#asterisk TimRiker (i=timr@bzflag/projectlead/TimRiker) |
| 15:52.27 | *** join/#asterisk nny_1 (n=scott@64.203.244.146) |
| 15:52.29 | BrianR___ | [TK]D-Fender: heh... It's not my job anymore, so I don't have to worry about it... |
| 15:52.32 | [TK]D-Fender | TimRiker: Many greetings :) |
| 15:52.33 | mmlj4 | thanks for the promotion, but SIP has occasional problems with NAT |
| 15:52.34 | mort_gib | mmlj4: Users like them... |
| 15:52.41 | nny_1 | i think i have the making of a daily WTF here |
| 15:52.55 | mmlj4 | oh, I loathe polycom, too |
| 15:52.57 | nny_1 | another Monday, and IAX between two hosts doesn't work |
| 15:53.00 | mmlj4 | linksys for me, man |
| 15:53.01 | TimRiker | waves |
| 15:53.08 | stevetotaro | ditch IAX2 |
| 15:53.08 | nny_1 | yet pingpoker shows port etc as fine/up |
| 15:53.11 | nny_1 | heh |
| 15:53.14 | nny_1 | tempting |
| 15:53.22 | stevetotaro | seriously |
| 15:53.28 | mmlj4 | stevetotaro: why? |
| 15:53.28 | nny_1 | yeah i may just do that |
| 15:53.37 | mmlj4 | what's wrong with it? |
| 15:53.42 | stevetotaro | i have fixed countless issues by kicking out IAX2 for SIP |
| 15:53.54 | stevetotaro | two large ITSPs |
| 15:54.08 | stevetotaro | i was a "hero" |
| 15:54.14 | TimRiker | ops: freenode per-account limits are sometimes causing apt/ibot/infobot/jbot/purl issues. I'm planning to drop the jbot nick from active use. Any preference on which other nick you want in here? |
| 15:54.47 | mmlj4 | enopurl *snif* |
| 15:54.47 | stevetotaro | ~jbot |
| 15:54.48 | jbot | well, jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch, or suck, or a pain in the ass |
| 15:55.06 | Maliuta | we love our jbot |
| 15:55.06 | nny_1 | nominates [TK]D-Fendersbitch |
| 15:56.05 | drmessano | I think [TK]D-Fender should name it.. its his baby |
| 15:56.23 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 15:56.42 | nny_1 | i would just switch it to SIP, but i am a masochist and would love to know why it isn't working, on Monday twice now, when network stuff (port access, etc) seems normal |
| 15:56.56 | nny_1 | i switched the port this morning to no avail |
| 15:57.18 | [TK]D-Fender | Mah bitch! |
| 15:57.37 | drmessano | I would vote for AsterBitch |
| 15:57.41 | stevetotaro | nny, cannot really help you there, i go with what works and drop what doesn't |
| 15:57.45 | nny_1 | DFB |
| 15:57.59 | nny_1 | or TKDFB |
| 15:58.00 | stevetotaro | except these damn cigarettes |
| 15:58.02 | Maliuta | AsterKiller |
| 15:58.16 | [TK]D-Fender | TimRiker: I would say AstBot makes sense |
| 15:58.16 | stevetotaro | JTwat |
| 15:58.19 | Maliuta | Astalker? |
| 15:58.31 | drmessano | AstBot is good |
| 15:58.42 | drmessano | AstMaster works too |
| 15:58.44 | Maliuta | mourns jbot |
| 15:58.48 | [TK]D-Fender | Should be short, and meaningful |
| 15:58.53 | jstew | astmaster sounds like a porn |
| 15:58.58 | nny_1 | ha |
| 15:59.04 | drmessano | PBXPr0n |
| 15:59.08 | jstew | lol |
| 15:59.09 | Maliuta | yes! |
| 15:59.22 | drmessano | Open your dialplan, baby.. I got a call for you |
| 15:59.26 | TimRiker | the choices are apt, ibot, infobot, or purl. it's a freenode nick limit issue. |
| 15:59.38 | Maliuta | is that a trunk in your pocket? ... |
| 15:59.39 | drmessano | ah |
| 15:59.43 | hi365 | anyone using an audiocodes mp202 + fxo port? |
| 15:59.55 | drmessano | ibot sounds the best |
| 16:00.05 | Maliuta | it's jbot-- |
| 16:00.08 | *** join/#asterisk digitalirony (i=digitali@my.grandma.uses.shellium.org) |
| 16:00.16 | [TK]D-Fender | TimRiker: Honestly the name matters little as long as it works :) that in mind I'd tryust your judgement. |
| 16:00.20 | nny_1 | wonders how many times TimRiker has to do this today.. :) |
| 16:00.28 | drmessano | ibot is close enough to jbot and makes it easy |
| 16:01.06 | stevetotaro | ibot is trademarked |
| 16:01.31 | [TK]D-Fender | abot! |
| 16:01.37 | Maliuta | thebot! |
| 16:01.43 | [TK]D-Fender | heads to lunch, BBIAB |
| 16:01.47 | Maliuta | rowbot! |
| 16:01.48 | stevetotaro | dabot |
| 16:02.16 | TimRiker | nny_1: about 20 times. |
| 16:02.29 | nny_1 | TimRiker: heh :) |
| 16:02.36 | TimRiker | jbot: chanset #asterisk autojoin infobot |
| 16:02.36 | jbot | Setting autojoin for #asterisk to 'infobot'; was 'jbot'. |
| 16:02.39 | *** join/#asterisk CunningPike (n=CunningP@204.239.10.119) |
| 16:02.46 | TimRiker | jbot: part #asterisk |
| 16:02.46 | *** part/#asterisk jbot (i=ibot@rikers.org) |
| 16:02.59 | *** join/#asterisk infobot (i=ibot@rikers.org) |
| 16:02.59 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.6 (2009/02/23), 1.4.24 (2009/03/16), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.3 (2009/02/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #asterisk-commits |
| 16:03.05 | stevetotaro | ~infobott |
| 16:03.07 | TimRiker | infobot: save |
| 16:03.07 | infobot | saved user and chan files |
| 16:03.11 | stevetotaro | ~infobot |
| 16:03.12 | infobot | extra, extra, read all about it, infobot is [infobot], or infobot, or likes abuse |
| 16:03.42 | TimRiker | same bot, different name. still logs publicly. |
| 16:03.44 | TimRiker | ~logs |
| 16:03.45 | infobot | All conversations are logged to http://ibot.rikers.org/channel, where "channel" is replaced by the URL-encoded channel name, such as %23freenode for #freenode. Lines starting with spaces are not logged. |
| 16:03.47 | nny_1 | so SIP is an option over IAX, what's the argument for using IAX over it? Reading voip info about it, but article is not useful (go figure) |
| 16:03.56 | jjshoe | TimRiker as in tux screen tim riker? :P |
| 16:04.05 | TimRiker | jjshoe: the same. |
| 16:04.17 | drmessano | The famous one |
| 16:04.23 | jjshoe | TimRiker I have an un-molested tux screen I've never gotten around to molesting :P |
| 16:04.26 | drmessano | As opposed to the one that works at the CircleK |
| 16:04.41 | nny_1 | tuxscreen looks.. cool |
| 16:04.42 | Maliuta | ~book |
| 16:04.42 | infobot | [~book] 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://astbook.asteriskdocs.org or see ~buybook |
| 16:04.45 | drmessano | The CircleK guy knows NOTHING of Linux |
| 16:04.54 | jjshoe | nny_1 sell you one :P |
| 16:05.20 | nny_1 | :) |
| 16:05.27 | stevetotaro | nny, you don't use sip over iax you switch to sip from iax |
| 16:05.41 | jjshoe | no point in converting it now, since ata's are so cheap and easy |
| 16:05.42 | stevetotaro | you could of course use sip over vpn..... |
| 16:06.03 | nny_1 | stevetotaro: miscommunication, what i mean is one over the other, as in preference |
| 16:06.10 | drmessano | ... |
| 16:06.20 | jjshoe | TimRiker working on anything else fun you can talk about? |
| 16:07.40 | jjshoe | TimRiker holy.... are you still working with family search? arn't they the ones doing the distributed input of census records and what not? |
| 16:07.41 | TimRiker | jjshoe: trying hard to get another linux based smart cell phone to market. native sip capable. |
| 16:07.55 | TimRiker | jjshoe: guilty again. |
| 16:08.15 | TimRiker | 6000+ linux servers run the back end there. |
| 16:08.52 | jjshoe | TimRiker nice, I recently stumbled upon geni.com that got me into geneology... now I've been hunting out much better, and open, projects like it, but I would be lying if I said I didn't spend time on familysearch researching |
| 16:08.57 | jjshoe | TimRiker wow O.o |
| 16:09.19 | *** join/#asterisk jstoker (n=jstoker@88-107-192-59.dynamic.dsl.as9105.com) |
| 16:09.26 | TimRiker | this phone if/when it ships will be available with verizon with a data-only plan if you like, and sip capable. (video too) |
| 16:10.27 | jjshoe | nice |
| 16:11.03 | *** join/#asterisk sack (n=sack@49.Red-88-24-158.staticIP.rima-tde.net) |
| 16:11.19 | TimRiker | yeah. it will be if/when it happens. cellular companies are a pain to deal with. verizon less so that others from a device builder's perspective. |
| 16:11.28 | KavanS | does anyone here do video w/asterisk? |
| 16:11.59 | *** part/#asterisk jstoker (n=jstoker@88-107-192-59.dynamic.dsl.as9105.com) |
| 16:12.19 | phix | Engines at 1/4 impulse, engage! |
| 16:12.45 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.67) |
| 16:13.20 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:13.21 | stevetotaro | riker, lol |
| 16:13.31 | stevetotaro | number 1!!! |
| 16:14.17 | *** join/#asterisk kink0 (n=xchat@212.170.176.86) |
| 16:14.19 | kink0 | hello |
| 16:14.25 | kink0 | anyone using SS7 ? |
| 16:14.33 | stevetotaro | what has Jeff pulver been up to i wonder |
| 16:14.49 | stevetotaro | there are at least two ss7 mailing lists |
| 16:14.59 | stevetotaro | fairly active at stimes |
| 16:16.47 | *** part/#asterisk sack (n=sack@49.Red-88-24-158.staticIP.rima-tde.net) |
| 16:17.18 | kink0 | stevetotaro, ok, I will check it, I am a bit confused on this SS7 support... |
| 16:18.28 | *** join/#asterisk neverblue (n=jezus@unaffiliated/neverblue) |
| 16:18.54 | neverblue | where is voicemail configured, on the server side, not the client side? |
| 16:19.05 | phix | stevetotaro: :) nah I am watching Voyger actually |
| 16:19.21 | phix | stevetotaro: she says it too |
| 16:19.34 | nny_1 | well, i'll ask a stupid question, where to start with Unable to create channel of type 'IAX2' (cause 3 - No route to destination) |
| 16:19.41 | nny_1 | only form one side (remote) |
| 16:19.43 | nny_1 | from |
| 16:19.50 | nny_1 | pingpoker says port on local is available |
| 16:19.51 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 16:19.53 | mort_gib | nny_1: DNS and traceroute |
| 16:20.10 | stevetotaro | what does your dial statement say |
| 16:20.11 | kink0 | my question about SS7 is if Asterisk manages mtp3 , or would require some external softswitch to do that. |
| 16:20.15 | stevetotaro | dial/iax2? |
| 16:20.25 | nny_1 | not using FQDN for hosts, and i can ssh to local from remote and pingpoker perl script says 4569 on local |
| 16:20.42 | neverblue | i looked at voicemail.conf, but I don't see what I need, for example, * is using /var/spool/asterisk to put the voicemails into a directory, but I want to adjust those settings (its creating a .WAV and a .wav file, which is something I would like to repair). Where would I adjust those settings? |
| 16:20.55 | nny_1 | stevetotaro: Dial/PEER/EXTEN |
| 16:21.14 | *** join/#asterisk lbt_ (n=david@78.32.229.233) |
| 16:21.17 | nny_1 | er Dial (IAX2/PEER/EXTEN) |
| 16:21.25 | *** part/#asterisk TimRiker (i=timr@bzflag/projectlead/TimRiker) |
| 16:21.29 | stevetotaro | drop the 2 |
| 16:22.47 | *** part/#asterisk lbt_ (n=david@78.32.229.233) |
| 16:22.49 | nny_1 | <PROTECTED> |
| 16:22.59 | stevetotaro | or your peer is not defined correctly |
| 16:23.21 | stevetotaro | i have a stupid freakin virus that has jacked my google searches |
| 16:24.02 | *** join/#asterisk jstoker (n=jstoker@88-107-192-59.dynamic.dsl.as9105.com) |
| 16:24.11 | jameswf | whats a virus... |
| 16:24.15 | nny_1 | stevetotaro: i know how to fix that |
| 16:24.22 | jameswf | oh thats a windows thing |
| 16:24.27 | nny_1 | go to My Computer, right click manage |
| 16:24.33 | stevetotaro | it is firefox thing |
| 16:24.47 | jermey_g | an asterisk box used to live behind a wifi AP, one day an incoming call on its sip interface came which landed into the right extension. That extension was there to read dtmf digits from every incoming caller. When the call was coming from a remote sip proxy, no digits could be read while if a friend calls in, its dtmf digits could be read. Everyone knew the dtmf mode in use was out of band rfc2833. but still some how from re |
| 16:24.52 | Qwell | RIP jbot :( |
| 16:25.01 | nny_1 | usually there is a driver installed in system devices that is redirecting searches, but either way |
| 16:25.04 | jermey_g | can a wifi AP block out of band dtmf |
| 16:25.10 | jameswf | someone killed jbot |
| 16:25.18 | Qwell | jameswf: renamed |
| 16:25.26 | Qwell | points to infobot |
| 16:25.34 | stevetotaro | ~infobot |
| 16:25.35 | infobot | [infobot] [infobot], or infobot, or likes abuse |
| 16:25.41 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 16:25.46 | jameswf | ~twitter |
| 16:25.47 | nny_1 | stevetotaro: http://pastebin.com/m2f824d36 |
| 16:25.51 | jameswf | eh |
| 16:26.25 | *** part/#asterisk jstoker (n=jstoker@88-107-192-59.dynamic.dsl.as9105.com) |
| 16:26.25 | nny_1 | stevetotaro: if IE does the same thing (searches google, but when you click the link you get wierd URL with go in the URL, lemme know, i have removed that bastard countless times |
| 16:26.28 | stevetotaro | nny, try this first |
| 16:27.02 | stevetotaro | dial(IAX2/username:pass@IP) |
| 16:27.35 | nny_1 | hmm |
| 16:27.38 | nny_1 | just saw this in console |
| 16:27.39 | nny_1 | Rejected connect attempt from 64.203.239.92, request '100@from-iax' does not exist |
| 16:27.50 | nny_1 | dp typo somewhere? |
| 16:28.24 | stevetotaro | so there should be a context named from-iax in extensions.conf with an exten 100 or pattern match |
| 16:28.33 | nny_1 | nm |
| 16:28.38 | nny_1 | someone dialing wrong >< |
| 16:28.47 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:28.53 | *** join/#asterisk propellerhead (n=yogurt2u@host110.190-136-239.telecom.net.ar) |
| 16:29.00 | nny_1 | actually |
| 16:29.08 | nny_1 | the context is from=IAX2 >< |
| 16:29.14 | nny_1 | er from-IAX2 |
| 16:29.30 | [TK]D-Fender | ~pb |
| 16:29.31 | 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/ |
| 16:29.38 | [TK]D-Fender | infobot is is.... |
| 16:29.41 | [TK]D-Fender | it* |
| 16:30.11 | stevetotaro | someone customize infobot to be a bit more friendly like jbot was |
| 16:30.27 | stevetotaro | some personality |
| 16:30.39 | nny_1 | stevetotaro: i think i found it.. to some degree.. the person who originally set this up has a context error |
| 16:31.02 | *** join/#asterisk mog (n=mog@c-68-62-169-246.hsd1.al.comcast.net) |
| 16:31.02 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:31.11 | stevetotaro | anything can be taken out of context, wriite that down.... |
| 16:31.20 | nny_1 | heh |
| 16:31.58 | jermey_g | which of my ports are blocked? how do i know |
| 16:32.03 | nny_1 | had peer-context and from context as from-iax in both iax.conf, but in remote host had the context named from-IAX2 |
| 16:32.03 | jermey_g | my box cant get the dtmf |
| 16:32.10 | nny_1 | however it still doesn't work |
| 16:32.59 | stevetotaro | just stop typing into irc, look at what is in the dialplan, turn up verbose and make test calls |
| 16:33.06 | stevetotaro | the answer will jump out at you |
| 16:33.37 | nny_1 | stevetotaro: ok, but fwiw it worked all last week, after stopping monday, and here i am again, so I am not sure what part of this is dynamic |
| 16:33.48 | axisys | can I add a second name and email in voicemail.conf file for same number? |
| 16:33.58 | stevetotaro | nny, go to sip |
| 16:34.01 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 16:34.03 | stevetotaro | save yourself |
| 16:34.19 | stevetotaro | if it worked and now it doesn't, then something has changed |
| 16:34.26 | nny_1 | stevetotaro: yes |
| 16:34.34 | axisys | 8720 => 1234,Asif Iqbal,vadud3@gmail.com <-- can I add another entry with 8720 with second name and email? |
| 16:34.36 | nny_1 | stevetotaro: te question is what, only i have access to this system now |
| 16:35.11 | stevetotaro | that is the million dollar question, what has changed |
| 16:35.17 | stevetotaro | what if you reload |
| 16:35.27 | stevetotaro | can you stop asterisk? |
| 16:35.33 | stevetotaro | maybe reboot |
| 16:35.46 | stevetotaro | shouldn't have to but you never know |
| 16:37.34 | *** join/#asterisk infobot (i=ibot@rikers.org) |
| 16:37.34 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.6 (2009/02/23), 1.4.24 (2009/03/16), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.3 (2009/02/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #asterisk-commits |
| 16:39.11 | neverblue | nevermind :D |
| 16:39.19 | neverblue | i think its set at compile time |
| 16:39.23 | jermey_g | adds vadud3@gmail.com to spam bomber |
| 16:39.23 | nny_1 | stevetotaro: reloaded multiple times |
| 16:39.40 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 16:40.11 | nny_1 | eh. I'll save some money and time, if anyone wants to assist with this issue directly i can PP you for your time |
| 16:40.39 | *** join/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 16:47.49 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 16:48.37 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 16:49.08 | nny_1 | odd |
| 16:49.17 | [TK]D-Fender | even |
| 16:49.28 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 16:49.30 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 16:49.30 | nny_1 | i have Rejected connect attempt from 64.203.239.92, request '108@from-iax' does not exist in console, but that context is no where in /etc/asterisk/* (ala grep) |
| 16:49.42 | nny_1 | heh |
| 16:50.07 | [TK]D-Fender | nny_1: Its probably being called fmo the remote side. You know they can target contexts in THEIR dial, right? |
| 16:50.16 | [TK]D-Fender | from* |
| 16:50.32 | nny_1 | yeah i checked both side |
| 16:50.35 | nny_1 | sides* |
| 16:50.50 | telnettech | ok TK im going to fustrate you with my question....i will try to explain it as best as possible |
| 16:50.55 | telnettech | http://pastebin.com/d3d8090d8 |
| 16:51.38 | telnettech | I want to take the caller id and if it begins with '8' and is a 4 digit extension number, I need to send it to a macro that goes to voicemail |
| 16:52.06 | nny_1 | [TK]D-Fender: pm inc |
| 16:52.14 | telnettech | but it the callerid begins with '8' and is 3 digits, i want it to go to the current macro for voicemail |
| 17:00.07 | *** join/#asterisk jstoker (n=jstoker@88-107-192-59.dynamic.dsl.as9105.com) |
| 17:01.43 | telnettech | anything |
| 17:02.06 | *** part/#asterisk jstoker (n=jstoker@88-107-192-59.dynamic.dsl.as9105.com) |
| 17:02.26 | telnettech | anybody? |
| 17:03.31 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 17:05.32 | telnettech | can anybody help me with this request? |
| 17:07.11 | jermey_g | [TK]D-Fender:any idea on debugging a dtmf sent from phone to the sip proxy via a service provider which supports out of band rfc2833? |
| 17:07.18 | [TK]D-Fender | telnettech: http://pastebin.com/m42316629 |
| 17:07.57 | telnettech | Thanks TK |
| 17:08.31 | workdraft | rtp proxy is only needed for voip server and voip clients are behind different NAT, right? |
| 17:08.58 | [TK]D-Fender | telnettech: You are using long deprecated variables and really need to read CHANNELVARIABLES again and brush up on your expressions. |
| 17:09.06 | [TK]D-Fender | workdraft: No. |
| 17:09.20 | telnettech | i will TK |
| 17:09.23 | telnettech | thanks again |
| 17:10.45 | jermey_g | anybody |
| 17:10.46 | telnettech | TK: as i have said before, I have to use what our developers have given us......they are supposed to have a 1.4.24 version ready for production by the end of this week, but there is no plan to update the 20 or 30 systems with 1.2 * already |
| 17:10.58 | [TK]D-Fender | ~sipnat |
| 17:10.59 | 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 |
| 17:11.03 | [TK]D-Fender | workdraft: READ ^^^ |
| 17:11.22 | [TK]D-Fender | telnettech: Yes, but you were using 1.0 syntax... |
| 17:11.32 | workdraft | ok thnx. |
| 17:11.34 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 17:12.30 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 17:16.20 | *** join/#asterisk tecnico (n=tecnico@75.76.169.148) |
| 17:17.08 | jermey_g | fuckers! |
| 17:17.47 | drmessano | ? |
| 17:19.01 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 17:19.04 | *** join/#asterisk MindTheGap (n=MindTheG@189.59.134.145) |
| 17:20.48 | MaliutaLap | jermey_g: where? pics? ;) |
| 17:23.30 | guax | hah |
| 17:24.09 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 17:26.41 | *** join/#asterisk w9sh (n=sph@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 17:27.28 | *** join/#asterisk `paul (n=kutimoy@121.97.99.151) |
| 17:28.24 | `paul | does FWD still allow call to TF numbers for free? |
| 17:29.11 | MaliutaLap | shite 03:30, later than I thought it was |
| 17:29.18 | MaliutaLap | goes to bed |
| 17:30.04 | *** join/#asterisk jblack (n=jblack@pool-71-173-9-11.sctnpa.east.verizon.net) |
| 17:32.20 | drmessano | `paul: many others do that |
| 17:33.57 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 17:33.58 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 17:40.00 | *** join/#asterisk CunningPike (n=CunningP@204.239.10.119) |
| 17:40.43 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 17:44.25 | *** join/#asterisk DavidR2008 (n=chatzill@c-68-34-116-180.hsd1.md.comcast.net) |
| 17:45.20 | DavidR2008 | quick question: is there a way to have sip debug output sent to a file? |
| 17:47.11 | *** join/#asterisk bbryant (n=Brett_Br@209.192.100.193) |
| 17:54.34 | drmessano | Paul you can message me here |
| 17:56.33 | *** join/#asterisk saftsack (n=saftsack@p57924F5F.dip.t-dialin.net) |
| 17:56.38 | Qwell | ~init |
| 17:56.39 | infobot | hmm... init is a function that calls all the other subsystems, net, mm, mtd, fs's, mount rootfs, etc |
| 17:57.06 | `paul | :) |
| 17:57.07 | Qwell | ~shutdown |
| 17:57.34 | `paul | do you know alternatives to FWD? (for doing free outbound calls to US tf numbers) |
| 17:58.39 | jblack | Oh come on. Can't you afford the one penny a minute that is available on the market? |
| 17:58.54 | jjshoe | Qwell wanna give me a hug? |
| 17:59.05 | jjshoe | Qwell I bought two lottery tickets, if I win you can be my pool boy. |
| 17:59.50 | Qwell | jjshoe: fix this recording bug for me |
| 18:01.15 | telnettech | all....what is the best way to troubleshoot and resolve echo issues with IP phones? |
| 18:01.27 | trelane | `paul: 4WD! |
| 18:01.31 | *** join/#asterisk Spirits-Sight (n=christop@c-76-118-227-40.hsd1.ma.comcast.net) |
| 18:01.42 | trelane | has it on his jeep, goes long distances fine! |
| 18:02.34 | jjshoe | Qwell link? |
| 18:02.41 | Qwell | jjshoe: ask Ethan |
| 18:02.48 | jjshoe | Qwell oh, a permissions thing? |
| 18:02.53 | Qwell | permissions on recordings are broken because FreePBX never fixes it |
| 18:03.05 | Qwell | need to chmod after the copy |
| 18:03.09 | jjshoe | Qwell oh, is this the fing bug you and I were talking about the other day? umask or something? |
| 18:03.19 | Spirits-Sight | Just wondering, if I got my port forwards setup right. SIP 5060-5060 UDP at the ip of * and RTF 10000-20000 UDP at ip address of * are these right for a SIP setup? do or should I put any other one in to it? |
| 18:03.25 | Qwell | dunno. it's a simple fix though |
| 18:03.26 | drmessano | `paul: MOST ITSPs pass free TF calls at no charge.. you can also send them to voiper.ipkall.com |
| 18:04.24 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 18:04.55 | `paul | thanks |
| 18:05.56 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 18:08.51 | *** join/#asterisk VoipForces (n=kvirc@firewall.privalodc.com) |
| 18:09.02 | *** join/#asterisk dieguito84 (n=diego@host19-194-dynamic.12-79-r.retail.telecomitalia.it) |
| 18:13.07 | *** join/#asterisk Spirits-Sight (n=christop@c-76-118-227-40.hsd1.ma.comcast.net) |
| 18:17.10 | Spirits-Sight | ok, got a question or two: First how does this all work if there is two device that use SIP on the same router? I have my * box and phone plug into the same router, then I have another device called video phone that uses SIP also <-- works with a real phone number (for the deaf/hoh) which is connected to the IP address (public) so when person calls the phone number the device rings there be it a hearing or deaf person. any h |
| 18:17.42 | *** join/#asterisk jstoker (n=jstoker@88-107-192-59.dynamic.dsl.as9105.com) |
| 18:18.22 | *** part/#asterisk jstoker (n=jstoker@88-107-192-59.dynamic.dsl.as9105.com) |
| 18:20.15 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 18:24.02 | *** join/#asterisk j_kroon (n=jkroon@dsl-240-144-26.telkomadsl.co.za) |
| 18:29.15 | *** join/#asterisk CunningPike (n=CunningP@204.239.10.119) |
| 18:29.37 | *** join/#asterisk propellerhead (n=yogurt2u@155-38-17-190.fibertel.com.ar) |
| 18:30.25 | *** join/#asterisk CunningPike (n=CunningP@204.239.10.119) |
| 18:32.57 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-57a67221e6a2a566) |
| 18:33.51 | kink0 | there some specific channel about SS7 or SIGTRAN ? |
| 18:37.29 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 18:38.19 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 18:39.36 | Linuturk | is a new dCAP |
| 18:39.41 | Linuturk | as of today! |
| 18:40.02 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:40.24 | rbd | does the Dial() command handle blind transfers (e.g. SIP REFER, get it off the box), or is Transfer() required for that? |
| 18:41.35 | jjshoe | Qwell so a friend of mine just tried to get out his verizon contract and failed big time, I know, he must be weaksauce |
| 18:42.05 | jjshoe | Qwell time to start a business where for a portion of what people have to pay to get out of contracts I can get them out, and if I don't I issue a refund... :D |
| 18:42.17 | drmessano | Verizon doesnt have cancellation fees anymore |
| 18:42.26 | drmessano | We got that letter over a year ago |
| 18:43.22 | drmessano | How lame |
| 18:43.33 | jjshoe | for cell phones? |
| 18:43.36 | [TK]D-Fender | rbd: blind transfer from a phone does not tell * to send the call somewhere outside of *'s responsibility. It only says to release the phon end from the call and have it process on its own. To have that call choose to attempt a complete hand-off you would use "transfer" in the dialplan called by the exten you transfer it to. |
| 18:43.44 | drmessano | jjshoe: yes |
| 18:43.53 | drmessano | I may have even saved the letter |
| 18:44.08 | drmessano | Seems like that was mandated by some higher authority |
| 18:44.16 | drmessano | New legislation perhaps |
| 18:44.20 | jjshoe | no, they offer month month crap, which I presume is just pay as you go stuff |
| 18:44.30 | jjshoe | drmessano this is what I ran into: http://www.boston.com/business/personalfinance/articles/2008/09/23/verizon_wireless_plans_no_termination_fee_pact/ |
| 18:44.32 | drmessano | Not what I am talking about |
| 18:44.56 | drmessano | Ah |
| 18:45.03 | drmessano | Right |
| 18:45.18 | drmessano | Its not pay as you go |
| 18:45.30 | drmessano | Its month-month, no contract |
| 18:45.54 | drmessano | They have "contracts" but they're non-binding, and more like lock in for specific rate plans |
| 18:46.18 | jjshoe | no matter, either way, I'm good at getting out of crap like that, I mearly said something to Qwell because I know he's good at it too |
| 18:46.21 | drmessano | Monthly subscribers will pay the same rates as those with long-term contracts, though they won't get discounts on new handsets, Verizon Wireless spokeswoman Brenda Raney said yesterday. |
| 18:46.39 | drmessano | If you stay in a "Contract", its like a customer loyalty thing |
| 18:46.46 | drmessano | But theres nothing to get out of! |
| 18:46.59 | drmessano | You just call them and tell them you want out, and they shut it off |
| 18:47.14 | jjshoe | I got out of my verizon contract when the iphone came out because they changed their rates to calling mexico |
| 18:47.18 | drmessano | Unless hes a dumbass and they gave him the AOL / DirecTV pitch |
| 18:47.21 | jjshoe | I only had to talk to five people to get out, but out I did. |
| 18:47.47 | drmessano | um ok |
| 18:50.20 | jaytee | I wish I was a mult-billionaire so I could offer some impoverished town somewhere in America an absurd amount of money with the only condition being that they change the town's name and all signs referring to it as Douchebag. |
| 18:50.25 | drmessano | A lot of companies like AOL, DirecTV, Comcast, etc have teams that in place to convince you to stay with them by throwing incentives and discounts at the customer.. if anything, thats what you would run into, because there is no contract to break |
| 18:50.42 | drmessano | and nothing you have to "talk them into" or "talk out of" |
| 18:51.14 | drmessano | You can go to month-month or nothing at all with one call |
| 18:52.59 | *** join/#asterisk Notre1 (n=Notre1@208.72.133.110) |
| 18:53.21 | Spirits-Sight | ok, got a question or two: First how does this all work if there is two device that use SIP on the same router? I have my * box and phone plug into the same router, then I have another device called video phone that uses SIP also <-- works with a real phone number (for the deaf/hoh) which is connected to the IP address (public) so when person calls the phone number the device rings there be it a hearing or deaf person. any h |
| 18:53.49 | jjshoe | drmessano ? |
| 18:53.54 | jjshoe | jaytee sounds like fun to me :P |
| 18:54.00 | *** join/#asterisk _brent_ (n=_brent_@jive.fttp.xmission.com) |
| 18:54.29 | jaytee | "hey, where you from?" "I'm from Douchebag!" "Oh! The one in Idaho or the one in Arkansas?" |
| 18:54.34 | jjshoe | jaytee i believe there is a town for sale on ebay currently, haven't looked in a while though |
| 18:54.54 | jjshoe | jaytee there's a town, and I can't recall how it's spelled, but it's pronounced "my anus", it was on an episode of jackass |
| 18:55.06 | jaytee | I don't want to buy the town, I just want to bribe them with money to change the name. |
| 18:55.14 | jjshoe | Mianus |
| 18:55.23 | jjshoe | http://www.youtube.com/watch?v=9Fytp3c5jEA nsfw |
| 18:55.45 | jaytee | there's a town on I-5 just north of Mt Shasta in Northern California called Weed |
| 18:55.53 | nny_1 | Spirits-Sight depends |
| 18:56.09 | jaytee | there's a highway sign saying, "Weed, next 3 exits" |
| 18:56.18 | Notre1 | Can I allow access to the MeetMe admin menu via the * key w/o asking everyone coming into confernce for a pin? I would like to just use adminpin without userpin. |
| 18:56.19 | nny_1 | Spirits-Sight: usually different sip auth ports and rtp ports |
| 18:56.25 | nny_1 | jaytee holy crap driven by that |
| 18:56.35 | jaytee | funny ain't it? |
| 18:56.47 | jaytee | and I've even been to Boring, Oregon |
| 18:56.48 | nny_1 | jaytee in Oregon, delivering furniture to a Harrahs from a jewish caprenter |
| 18:56.59 | nny_1 | jaytee: heh yeah |
| 18:57.07 | nny_1 | we always wondered what was better, north weed or south weed |
| 18:57.10 | jaytee | I used to live in Medford |
| 18:57.27 | nny_1 | i lived in southern cal, was up that way for the job. |
| 18:57.32 | nny_1 | shasta was awesome at 7 am heh |
| 18:57.33 | jaytee | which is just over the Cali border in Oregon |
| 18:57.33 | Spirits-Sight | nny_1: ok, so could you tell me what I should do, I think I would have more control of the * then the other device as the other device uses a thrid party software / hardware |
| 18:57.57 | jjshoe | anyone who's done la to vegas gets to seen a fun sign |
| 18:57.58 | jaytee | Shasta's a really pretty area and the lake area is great for camping. |
| 18:58.06 | jjshoe | i've seen it, think qwell's sent me a link before or someone |
| 18:58.06 | nny_1 | Spirits-Sight: just move sip auth port to non standard (5065 or something) |
| 18:58.10 | jjshoe | it's like zzyzx or something |
| 18:58.14 | nny_1 | yeah |
| 18:58.19 | nny_1 | passed by that one many times |
| 18:58.20 | jjshoe | http://en.wikipedia.org/wiki/Zzyzx,_California |
| 18:58.22 | nny_1 | zzyzx road |
| 18:59.01 | jjshoe | nny_1 last time I went to vegas I stopped at lake delores and snapped up some nice pictures |
| 18:59.10 | nny_1 | the giant dinosaurs between so cal and vegas are cool too |
| 18:59.16 | jjshoe | nny_1 giant dinosaurs? |
| 18:59.19 | nny_1 | i believe they are the same ones as in pee wee's big adventure |
| 18:59.32 | jaytee | when i was a kid we were making prank calls and we called the very last name in the phone book. the guy's name was Zwig Zizzy. He answered and even though he knew it was a prank call he was really cool about it. |
| 18:59.35 | jjshoe | nny_1 I've never seen them... wtf? what city? |
| 18:59.49 | Spirits-Sight | nny_1: how and where do I make these changes? I know I would in my router do portforward and make a new name say like asterick-sip 5065-5065 udf but where else do I make the change |
| 18:59.53 | jaytee | "Everybody's got a big but" |
| 19:00.19 | nny_1 | http://en.wikipedia.org/wiki/Cabazon,_California |
| 19:00.41 | nny_1 | http://images.google.com/images?client=firefox-a&rls=org.mozilla:en-US:official&channel=s&hl=en&q=cabazon%20dinosaurs&um=1&ie=UTF-8&sa=N&tab=wi |
| 19:01.18 | *** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) |
| 19:01.20 | jjshoe | nny_1 ahh ok, wasn't on my way to vegas, I was thinking more along the lines with out at baker or something |
| 19:01.36 | nny_1 | Spirits-Sight: check voip-info.org for the proper sip parameters in sip.conf |
| 19:01.41 | nny_1 | and |
| 19:01.42 | nny_1 | ~book |
| 19:01.43 | infobot | [~book] 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://astbook.asteriskdocs.org or see ~buybook |
| 19:01.52 | jaytee | "Dinosaurs are just a bunch o' malarkey made up by atheist scientists to confuse people and lure them away from Jesus!" (ducks and runs) |
| 19:02.01 | nny_1 | lol |
| 19:02.16 | Spirits-Sight | ok |
| 19:02.47 | Spirits-Sight | nny_1: do I need change any thing with the provider of sip (voicepulse)? |
| 19:02.55 | jjshoe | nny_1 http://photos-c.ak.fbcdn.net/photos-ak-sf2p/v363/168/21/215700850/n215700850_30995938_9188.jpg and http://photos-a.ak.fbcdn.net/photos-ak-sf2p/v363/168/21/215700850/n215700850_30995944_1038.jpg are probably my two best lake delores pictures |
| 19:03.42 | jjshoe | I got htose with a really long telephoto, yeah that's it :P |
| 19:04.13 | nny_1 | awesome |
| 19:04.33 | _brent_ | is there a realtime sip peer expert in the room? |
| 19:04.44 | _brent_ | or somebody that really knows their way around chan_sip.c? |
| 19:04.57 | jblack | hrmmm. Can someone familiar with the video codecs that windows supports msg me privately for an off topic question? |
| 19:05.11 | oej | _brent_ : Just ask your question |
| 19:05.17 | _brent_ | ok |
| 19:05.17 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-141-35.hrbgpa.dsl-w.verizon.net) |
| 19:05.25 | oej | Better than asking if someone can answer you... :-) |
| 19:05.34 | _brent_ | sure thing :-) |
| 19:06.20 | _brent_ | i've got several asterisk instances sharing load. peers are realtime. |
| 19:06.20 | _brent_ | rtcachefriends=yes |
| 19:06.20 | _brent_ | rtautoclear=no |
| 19:06.41 | _brent_ | i've got AIS set up for presence |
| 19:06.52 | _brent_ | presence works as long as rtautoclear=no |
| 19:07.56 | _brent_ | my problem is that if a peer registers on asterisk A, then later on asterisk B, when the registration expires on A, if a call for the peer arrives at A, it sees it as expired and sends congestion |
| 19:08.39 | _brent_ | if i set rtautoclear=yes|number the calls work again, but presence breaks as soon as it prunes the peer |
| 19:09.15 | oej | realtime and presence doesn't really work together |
| 19:09.24 | _brent_ | i think what i need is for the peer to not be pruned, but for the realtime stuff to be loaded again when a call comes for the peer |
| 19:09.34 | oej | Only if all the registrations is on the same server as all the subscriptions |
| 19:09.58 | _brent_ | with the ais module it works pretty well across multiple registrars |
| 19:10.09 | oej | Ah, you're using 1.6... |
| 19:10.11 | _brent_ | but breaks as soon as the first peer switches registrars |
| 19:10.12 | *** join/#asterisk manddy (n=moo@67.204.24.118) |
| 19:10.12 | _brent_ | yeah |
| 19:10.16 | Spirits-Sight | has any one used DD-WRT w/voip firmware with * |
| 19:10.21 | oej | Haven't looked at that yet. You need to ask someone else |
| 19:10.34 | _brent_ | k, anyone come to mind? |
| 19:10.39 | *** part/#asterisk manddy (n=moo@67.204.24.118) |
| 19:11.05 | oej | It's still very experimental, so I don't know if it's ready. I would not use it in production. |
| 19:11.22 | oej | The distributed event system that is |
| 19:11.57 | _brent_ | yeah, i'm just taking a look. it's promising. |
| 19:12.18 | _brent_ | oej: thanks for your help, by the way. i've come across answers in your old posts several times. |
| 19:12.22 | oej | Yes, we do need a distributed system |
| 19:12.22 | jblack | Ohhhh. I just had a horrible idea! |
| 19:15.18 | jblack | Worse. |
| 19:15.43 | jblack | oh, never mind. |
| 19:24.26 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 19:24.26 | *** join/#asterisk dlynes (n=daniel@CPE001617e008e3-CM00080d940644.cpe.net.cable.rogers.com) |
| 19:24.26 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 19:24.26 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-141-35.hrbgpa.dsl-w.verizon.net) [NETSPLIT VICTIM] |
| 19:24.26 | *** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) [NETSPLIT VICTIM] |
| 19:24.26 | *** join/#asterisk _brent_ (n=_brent_@jive.fttp.xmission.com) [NETSPLIT VICTIM] |
| 19:24.26 | *** join/#asterisk Notre1 (n=Notre1@208.72.133.110) [NETSPLIT VICTIM] |
| 19:24.26 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:24.26 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) [NETSPLIT VICTIM] |
| 19:24.26 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) [NETSPLIT VICTIM] |
| 19:24.26 | *** join/#asterisk CunningPike (n=CunningP@204.239.10.119) |
| 19:24.26 | *** join/#asterisk propellerhead (n=yogurt2u@155-38-17-190.fibertel.com.ar) [NETSPLIT VICTIM] |
| 19:24.26 | *** join/#asterisk j_kroon (n=jkroon@dsl-240-144-26.telkomadsl.co.za) [NETSPLIT VICTIM] |
| 19:24.27 | *** join/#asterisk Spirits-Sight (n=christop@c-76-118-227-40.hsd1.ma.comcast.net) [NETSPLIT VICTIM] |
| 19:24.27 | *** join/#asterisk dieguito84 (n=diego@host19-194-dynamic.12-79-r.retail.telecomitalia.it) [NETSPLIT VICTIM] |
| 19:24.27 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) [NETSPLIT VICTIM] |
| 19:24.27 | *** join/#asterisk bbryant (n=Brett_Br@209.192.100.193) |
| 19:24.27 | *** join/#asterisk DavidR2008 (n=chatzill@c-68-34-116-180.hsd1.md.comcast.net) [NETSPLIT VICTIM] |
| 19:24.27 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) [NETSPLIT VICTIM] |
| 19:24.27 | *** join/#asterisk `paul (n=kutimoy@121.97.99.151) [NETSPLIT VICTIM] |
| 19:24.27 | *** join/#asterisk w9sh (n=sph@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 19:24.27 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) [NETSPLIT VICTIM] |
| 19:24.27 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) [NETSPLIT VICTIM] |
| 19:24.27 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) [NETSPLIT VICTIM] |
| 19:24.27 | *** join/#asterisk shtoom (n=shtoom@121.246.167.147) [NETSPLIT VICTIM] |
| 19:24.27 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) [NETSPLIT VICTIM] |
| 19:24.27 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) [NETSPLIT VICTIM] |
| 19:24.28 | *** join/#asterisk mog (n=mog@c-68-62-169-246.hsd1.al.comcast.net) |
| 19:24.28 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 19:24.28 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 19:24.28 | *** join/#asterisk neverblue (n=jezus@unaffiliated/neverblue) [NETSPLIT VICTIM] |
| 19:24.28 | *** join/#asterisk kink0 (n=xchat@212.170.176.86) [NETSPLIT VICTIM] |
| 19:24.28 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.67) [NETSPLIT VICTIM] |
| 19:24.28 | *** join/#asterisk digitalirony (i=digitali@my.grandma.uses.shellium.org) [NETSPLIT VICTIM] |
| 19:24.28 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) [NETSPLIT VICTIM] |
| 19:24.28 | *** join/#asterisk workdraft (n=ian@122.52.117.21) [NETSPLIT VICTIM] |
| 19:24.28 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 19:24.28 | *** join/#asterisk jstew (n=jstewart@fw-ext.fusionary.com) [NETSPLIT VICTIM] |
| 19:24.28 | *** join/#asterisk jjshoe (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) [NETSPLIT VICTIM] |
| 19:24.28 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) [NETSPLIT VICTIM] |
| 19:24.28 | *** join/#asterisk hi365 (n=hi365@94.159.144.204) [NETSPLIT VICTIM] |
| 19:24.28 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 19:24.28 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) [NETSPLIT VICTIM] |
| 19:24.28 | *** mode/#asterisk [+ooo jtodd mog Deeewayne] by irc.freenode.net |
| 19:24.28 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) [NETSPLIT VICTIM] |
| 19:24.28 | *** join/#asterisk macros73_ (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net) [NETSPLIT VICTIM] |
| 19:24.28 | *** join/#asterisk jerlique (i=d208a802@gateway/web/ajax/mibbit.com/x-3bfb88aebc5ad1bd) [NETSPLIT VICTIM] |
| 19:24.28 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 19:24.28 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) [NETSPLIT VICTIM] |
| 19:24.28 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) [NETSPLIT VICTIM] |
| 19:24.28 | *** join/#asterisk moy (n=chatzill@74.12.124.89) |
| 19:24.28 | *** join/#asterisk dvdevel (n=devel@wiggum.digitalcoven.com) [NETSPLIT VICTIM] |
| 19:24.28 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) [NETSPLIT VICTIM] |
| 19:24.28 | *** join/#asterisk Badrobot- (n=Badrobot@cpe-76-173-233-75.socal.res.rr.com) [NETSPLIT VICTIM] |
| 19:24.29 | *** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca) [NETSPLIT VICTIM] |
| 19:24.29 | *** join/#asterisk Mw3 (i=mw3@mw3.hu) |
| 19:24.29 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) [NETSPLIT VICTIM] |
| 19:24.29 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 19:24.29 | *** join/#asterisk RobertLaptop (n=rmiddle@63.68.135.4) [NETSPLIT VICTIM] |
| 19:24.29 | *** join/#asterisk jplank (n=gbove@cpe-075-181-097-208.carolina.res.rr.com) [NETSPLIT VICTIM] |
| 19:24.29 | *** join/#asterisk path_ (n=path_@71-119-21-190.adsl.terra.cl) [NETSPLIT VICTIM] |
| 19:24.29 | *** join/#asterisk clintc (n=clintc@n128-227-7-200.xlate.ufl.edu) [NETSPLIT VICTIM] |
| 19:24.29 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) [NETSPLIT VICTIM] |
| 19:24.29 | *** join/#asterisk juanjoc (n=jcomella@200.69.219.113) |
| 19:24.29 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) [NETSPLIT VICTIM] |
| 19:24.29 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 19:24.29 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk beherit (n=albert@203.153.6.16) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk ltd (n=z@pat.transact.net.au) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk raden_work (n=jon@adsl-99-139-235-165.dsl.applwi.sbcglobal.net) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 19:24.30 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) |
| 19:24.30 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk _BBV_ (n=buklov@213.138.71.254) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk fiddur (i=fiddur@c042.rit.se) |
| 19:24.30 | *** join/#asterisk mandd (n=moo@67.204.24.118) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk BBHoss (n=bbhoss@72.146.26.249) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk awannabe (n=brad@ip70-162-201-97.ph.ph.cox.net) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk paulproteus (i=paulprot@2002:db69:2513:0:0:0:0:1) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk brian (n=brian@unaffiliated/brian) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk apeiron (n=apeiron@c-76-124-252-61.hsd1.pa.comcast.net) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk jackal (n=jackal@pool-98-118-240-229.clppva.fios.verizon.net) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk scooby2 (n=scooby2@pdpc/supporter/active/scooby2) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk bl4 (n=bl4qkuba@dsl5-ore-167.fiber.net) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk columbo (n=PeterFal@pool-98-119-2-125.lsanca.dsl-w.verizon.net) [NETSPLIT VICTIM] |
| 19:24.30 | *** join/#asterisk nix8n82 (n=nate@mo-65-41-196-62.sta.embarqhsd.net) [NETSPLIT VICTIM] |
| 19:24.30 | *** mode/#asterisk [+oo putnopvut denon] by irc.freenode.net |
| 19:24.31 | *** join/#asterisk andrewn (n=andrew@70.36.140.13) [NETSPLIT VICTIM] |
| 19:24.31 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) [NETSPLIT VICTIM] |
| 19:24.31 | *** join/#asterisk ph8 (i=ph8@unaffiliated/ph8) [NETSPLIT VICTIM] |
| 19:24.31 | *** join/#asterisk trask (n=trask@1407ds1-bav.0.fullrate.dk) [NETSPLIT VICTIM] |
| 19:24.31 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) [NETSPLIT VICTIM] |
| 19:24.31 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) [NETSPLIT VICTIM] |
| 19:24.56 | *** join/#asterisk Circlefusion (n=brian@74-132-114-73.dhcp.insightbb.com) [NETSPLIT VICTIM] |
| 19:24.56 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) [NETSPLIT VICTIM] |
| 19:24.56 | *** join/#asterisk VaGoNeTaS (n=debian@xen.datapartner.cl) |
| 19:24.56 | *** join/#asterisk wimt (i=wimt@freenode/staff/wikipedia.wimt) [NETSPLIT VICTIM] |
| 19:24.56 | *** join/#asterisk t (i=tom@freenode/staff/tomaw) |
| 19:24.56 | *** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca) [NETSPLIT VICTIM] |
| 19:24.56 | *** join/#asterisk LakeSolon (n=blake@64-83-211-119.dhcp.stcd.mn.charter.com) |
| 19:24.56 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) [NETSPLIT VICTIM] |
| 19:24.56 | *** join/#asterisk imm (n=ian@nv-76-3-138-164.dhcp.embarqhsd.net) [NETSPLIT VICTIM] |
| 19:24.56 | *** join/#asterisk quant0r (n=quant0r@quantorconsulting.com) [NETSPLIT VICTIM] |
| 19:24.56 | *** join/#asterisk scardinal (n=supreme@90.184.100.170) [NETSPLIT VICTIM] |
| 19:24.56 | *** join/#asterisk Failrar (n=Failrar@coffee.ipv6.kaufmann.tc) [NETSPLIT VICTIM] |
| 19:24.56 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) [NETSPLIT VICTIM] |
| 19:24.56 | *** join/#asterisk Tsion (n=scott@botters/staff/tsion) [NETSPLIT VICTIM] |
| 19:24.56 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) [NETSPLIT VICTIM] |
| 19:24.56 | *** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) [NETSPLIT VICTIM] |
| 19:24.56 | *** join/#asterisk macros73 (n=cs@c-71-61-74-104.hsd1.pa.comcast.net) [NETSPLIT VICTIM] |
| 19:24.56 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) [NETSPLIT VICTIM] |
| 19:24.56 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) [NETSPLIT VICTIM] |
| 19:24.56 | *** join/#asterisk _Steve_ (n=Steve@unaffiliated/steve/x-520345) [NETSPLIT VICTIM] |
| 19:24.56 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 19:24.56 | *** join/#asterisk kerframil (n=kerframi@gentoo/user/kerframil) |
| 19:24.56 | *** join/#asterisk phr3ak (n=phr3ak@gnet.hu) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-58-83.pskn.east.verizon.net) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk troy- (n=troy@38.103.146.115) |
| 19:24.58 | *** join/#asterisk rift0r (i=rift@420nugs.info) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk jermey_g (n=me@static-213-115-44-229.sme.bredbandsbolaget.se) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk hackeron (n=hackeron@gentoo/user/hackeron) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk NovceGuru (i=novcegur@server1.jsreedinc.com) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk x86 (n=x86@p3m/member/x86) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk stabler (n=seedbox@cpe-24-210-38-135.columbus.res.rr.com) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk ScriptFanix (i=vincent@2a01:e35:2f43:ae90:21a:70ff:fea3:44ab) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk Mark21 (n=mark@freenode/sponsor/mark17) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk tAnkOSX (i=tank@the.matrix.has-you.net) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk Woody2143 (n=Woody214@machine76.Level3.com) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk derrick_ (n=derrick@blinky-lights.org) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk M07w (n=M07w@216.234.116.122) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk Corydon76-dig (i=gray@pdpc/supporter/bronze/Corydon76-home) |
| 19:24.58 | *** join/#asterisk NoxIn- (n=noxin@zorlit.org) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 19:24.58 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk Daviey (n=Daviey@ubuntu/member/pdpc.gold.Daviey) |
| 19:24.58 | *** join/#asterisk tompaw (n=tompaw@slave20.tesserakt.eu) [NETSPLIT VICTIM] |
| 19:24.58 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 19:24.59 | *** join/#asterisk prg3 (n=prg3@playground.cein.ualberta.ca) [NETSPLIT VICTIM] |
| 19:24.59 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) [NETSPLIT VICTIM] |
| 19:24.59 | *** join/#asterisk nemik (n=cyanact@c-67-173-76-34.hsd1.il.comcast.net) [NETSPLIT VICTIM] |
| 19:24.59 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 19:24.59 | *** join/#asterisk elred (i=sauron@fucksheep.org) [NETSPLIT VICTIM] |
| 19:24.59 | *** join/#asterisk hardwire (n=hardwire@srv001.gandi.brutetech.com) [NETSPLIT VICTIM] |
| 19:24.59 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-f2afe336fea09726) [NETSPLIT VICTIM] |
| 19:24.59 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 19:24.59 | *** join/#asterisk slima (n=slima@unaffiliated/slima) [NETSPLIT VICTIM] |
| 19:24.59 | *** join/#asterisk swampwork (n=rew@64.238.252.218) [NETSPLIT VICTIM] |
| 19:24.59 | *** join/#asterisk BrianR___ (i=brianr@osiris.978.org) [NETSPLIT VICTIM] |
| 19:24.59 | *** join/#asterisk plundra (i=404@article.se) [NETSPLIT VICTIM] |
| 19:24.59 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 19:24.59 | *** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net) [NETSPLIT VICTIM] |
| 19:24.59 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 19:24.59 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) [NETSPLIT VICTIM] |
| 19:24.59 | *** join/#asterisk carrar (i=tim@osburn.com) [NETSPLIT VICTIM] |
| 19:24.59 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) [NETSPLIT VICTIM] |
| 19:24.59 | *** join/#asterisk japherwocky (n=japhy@li35-182.members.linode.com) [NETSPLIT VICTIM] |
| 19:24.59 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) [NETSPLIT VICTIM] |
| 19:25.00 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) [NETSPLIT VICTIM] |
| 19:25.00 | *** join/#asterisk McDarren (n=darren@shrubbery.chris.net.au) [NETSPLIT VICTIM] |
| 19:25.00 | *** join/#asterisk Takapa (i=vegard@junior.svanberg.no) |
| 19:25.00 | *** join/#asterisk festr_ (n=festr@ns.hiro.cz) |
| 19:25.00 | *** join/#asterisk trippss (n=sstoner@72.249.67.57) [NETSPLIT VICTIM] |
| 19:25.00 | *** join/#asterisk jksM (i=jks@193.189.93.254) [NETSPLIT VICTIM] |
| 19:25.00 | *** join/#asterisk emist (n=emist@unaffiliated/emist) [NETSPLIT VICTIM] |
| 19:25.00 | *** join/#asterisk Wayhigh (i=wayhigh@www.kevinlynn.com) [NETSPLIT VICTIM] |
| 19:25.00 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 19:25.00 | *** join/#asterisk bmg505 (n=leon@196-209-76-141-rndf-esr-5.dynamic.isadsl.co.za) [NETSPLIT VICTIM] |
| 19:25.00 | *** join/#asterisk brut- (n=brut-@h66-173-4-254.mntimn.dedicated.static.tds.net) [NETSPLIT VICTIM] |
| 19:25.00 | *** join/#asterisk fnordus (n=dnall@70.71.225.48) [NETSPLIT VICTIM] |
| 19:25.00 | *** join/#asterisk Lyma (n=Lyma@unaffiliated/lyma) [NETSPLIT VICTIM] |
| 19:25.00 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) [NETSPLIT VICTIM] |
| 19:25.00 | *** join/#asterisk snowboarder04 (n=un@serv.bemail.co.uk) [NETSPLIT VICTIM] |
| 19:25.00 | *** join/#asterisk elguero (n=elguero@ns1.nashuacs.com) |
| 19:25.00 | *** join/#asterisk Gary (i=gary@freenode/staff/colchester-lug.gary) |
| 19:25.00 | *** join/#asterisk inv_arp (n=junya@b07s03mr.corenetworks.net) [NETSPLIT VICTIM] |
| 19:25.00 | *** join/#asterisk prakriti (n=jeremy@97-92-218-156.dhcp.stls.mo.charter.com) [NETSPLIT VICTIM] |
| 19:25.00 | *** join/#asterisk dwayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 19:25.00 | *** join/#asterisk obruT (n=turbo@griffin.linux.hr) [NETSPLIT VICTIM] |
| 19:25.00 | *** mode/#asterisk [+oo Corydon76-dig Qwell] by irc.freenode.net |
| 19:25.00 | *** join/#asterisk mnicholson (n=mnichols@nat/digium/x-a38581c2af27997f) |
| 19:25.00 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) [NETSPLIT VICTIM] |
| 19:25.00 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1176207886.dsl.bell.ca) [NETSPLIT VICTIM] |
| 19:25.00 | *** join/#asterisk infinity1 (n=brendon@li6-32.members.linode.com) [NETSPLIT VICTIM] |
| 19:25.00 | *** join/#asterisk JT (n=j@unaffiliated/jt) [NETSPLIT VICTIM] |
| 19:25.01 | *** join/#asterisk roe (n=roe___@207-172-35-242.c3-0.eas-ubr15.atw-eas.pa.cable.rcn.com) [NETSPLIT VICTIM] |
| 19:25.01 | *** join/#asterisk p1mrx (n=paul@mediabox.apt.pmarks.net) [NETSPLIT VICTIM] |
| 19:25.01 | *** join/#asterisk [netman] (n=netman@142.Red-83-50-80.dynamicIP.rima-tde.net) [NETSPLIT VICTIM] |
| 19:25.01 | *** join/#asterisk inckie (i=kimse1@please.press.control-alt-delete.eu) [NETSPLIT VICTIM] |
| 19:25.01 | *** join/#asterisk Doc (n=scott@willers.employees.org) [NETSPLIT VICTIM] |
| 19:25.01 | *** join/#asterisk matt_ (n=matt@mattspc.ipv6.mattstone.net) [NETSPLIT VICTIM] |
| 19:25.01 | *** join/#asterisk CieD (n=ciemon@unaffiliated/cied) [NETSPLIT VICTIM] |
| 19:25.01 | *** join/#asterisk LeddyHM (n=NONE@mail.artica.net) [NETSPLIT VICTIM] |
| 19:25.01 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) [NETSPLIT VICTIM] |
| 19:25.01 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 19:25.01 | *** join/#asterisk saxa (n=sasa@host242-95-static.223-217-b.business.telecomitalia.it) [NETSPLIT VICTIM] |
| 19:25.01 | *** join/#asterisk LapTop006 (n=laptop00@gemini.chriskaine.com.au) [NETSPLIT VICTIM] |
| 19:25.01 | *** join/#asterisk thinko (n=jdoe6alp@smaug.rackdragon.com) [NETSPLIT VICTIM] |
| 19:25.01 | *** join/#asterisk the_5th_wheel (n=edd@webster.cybertek.co.za) [NETSPLIT VICTIM] |
| 19:25.01 | *** join/#asterisk unpaidbill (n=bill@alteredbeastiality.org) |
| 19:25.01 | *** join/#asterisk rajiv (n=rajiv@gentoo/developer/rajiv) [NETSPLIT VICTIM] |
| 19:25.01 | *** join/#asterisk trelane (i=trelane@funtoo/staff/trelane) [NETSPLIT VICTIM] |
| 19:25.02 | *** join/#asterisk gultig (n=dbriggs@70.96.32.114) [NETSPLIT VICTIM] |
| 19:25.02 | *** join/#asterisk Melifaro (n=melifaro@melifaro.astel.ru) [NETSPLIT VICTIM] |
| 19:25.02 | *** join/#asterisk pdmmm (i=pdm@pdm.sh) [NETSPLIT VICTIM] |
| 19:25.02 | *** join/#asterisk blackest_mamba (n=blackest@71.239.160.143) [NETSPLIT VICTIM] |
| 19:25.02 | *** join/#asterisk mbranca (n=matteo@2001:1418:130:0:21e:8cff:fe51:5b05) |
| 19:25.02 | *** join/#asterisk cy3o3 (n=cy@it.was.otherkids.net) |
| 19:25.02 | *** join/#asterisk keith4 (n=keith@lust.cc.lehigh.edu) [NETSPLIT VICTIM] |
| 19:25.02 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 19:25.02 | *** join/#asterisk Quintana (n=sylvain@aghnar.doowan.net) [NETSPLIT VICTIM] |
| 19:25.02 | *** join/#asterisk rbd (n=rbd@rrcs-96-10-27-206.se.biz.rr.com) |
| 19:25.02 | *** join/#asterisk vlt (n=dm@suez.activ-job.com) [NETSPLIT VICTIM] |
| 19:25.02 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) [NETSPLIT VICTIM] |
| 19:25.02 | *** join/#asterisk edoceo (n=edoceo@c-98-247-254-241.hsd1.wa.comcast.net) [NETSPLIT VICTIM] |
| 19:25.02 | *** join/#asterisk bisonit (n=randall@59.21.124.24.cm.sunflower.com) [NETSPLIT VICTIM] |
| 19:25.02 | *** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130) |
| 19:25.02 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) [NETSPLIT VICTIM] |
| 19:25.02 | *** join/#asterisk kn0x (n=pinochle@67.159.48.101) [NETSPLIT VICTIM] |
| 19:25.02 | *** join/#asterisk felipe_ (n=felipe@my.nada.kth.se) |
| 19:25.02 | *** join/#asterisk aiksa[LV] (n=aiks@mx.fiveplus.lv) [NETSPLIT VICTIM] |
| 19:25.02 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) [NETSPLIT VICTIM] |
| 19:25.02 | *** join/#asterisk postel (n=jp@wikimedia/Postel) [NETSPLIT VICTIM] |
| 19:25.02 | *** join/#asterisk viq (n=viq@unaffiliated/viq) [NETSPLIT VICTIM] |
| 19:25.02 | *** join/#asterisk slaney (n=slaney@67-207-131-109.slicehost.net) [NETSPLIT VICTIM] |
| 19:25.02 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 19:25.02 | *** join/#asterisk killfill (n=killfill@200.63.96.244) [NETSPLIT VICTIM] |
| 19:25.02 | *** join/#asterisk Talkradio (i=talkradi@linuxgeneration.ca) [NETSPLIT VICTIM] |
| 19:25.02 | *** join/#asterisk The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel) |
| 19:25.03 | *** mode/#asterisk [+o angler] by irc.freenode.net |
| 19:25.03 | *** join/#asterisk awk_r (n=awk_r@nat/digium/x-17edf8adec9acf8a) |
| 19:25.03 | *** join/#asterisk rashed2020_ (n=admin@67.205.245.208) [NETSPLIT VICTIM] |
| 19:25.03 | *** join/#asterisk hohum (n=dcorbe@206.71.169.115) [NETSPLIT VICTIM] |
| 19:25.03 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 19:25.03 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.com) [NETSPLIT VICTIM] |
| 19:25.03 | *** join/#asterisk De_Mon (i=de_mon@fl-67-77-166-5.dyn.embarqhsd.net) |
| 19:25.03 | *** join/#asterisk eharris (i=eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) [NETSPLIT VICTIM] |
| 19:25.03 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 19:25.03 | *** join/#asterisk defswork (n=andy@mx2.3gcomms.co.uk) [NETSPLIT VICTIM] |
| 19:25.03 | *** join/#asterisk ickmund (n=ickmund@ada-bcn-fw01.adamoeurope.com) [NETSPLIT VICTIM] |
| 19:25.04 | *** join/#asterisk lucasb (n=lucasb@office.telifon.com) [NETSPLIT VICTIM] |
| 19:25.04 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) [NETSPLIT VICTIM] |
| 19:25.04 | *** mode/#asterisk [+oo russellb file] by irc.freenode.net |
| 19:25.57 | tzafrir_laptop | ? |
| 19:25.57 | tzafrir_laptop | Asterisk is free software. It does not come with any warantee |
| 19:25.57 | Spirits-Sight | sorry supported |
| 19:25.57 | _brent_ | neither does most proprietary software... |
| 19:25.57 | tzafrir_laptop | DD-WRT is not a popular platform, and hence problems with it may be not as easy to reproduce |
| 19:25.57 | tzafrir_laptop | But apart from that, I don't see any inherent issue |
| 19:25.58 | dlynes | CunningPike: Good morning :) |
| 19:25.58 | Spirits-Sight | tzafrir_laptop: do you know of any way for me to load asterisk on to it |
| 19:25.58 | tzafrir_laptop | Spirits-Sight, I can't give you specific instructions. I know people use Asterisk with openwrt |
| 19:25.58 | tzafrir_laptop | And I'm not really closely familiar with those platforms |
| 19:25.58 | *** join/#asterisk bob_slacker (n=bob_slac@201.47.149.252) |
| 19:25.58 | tzafrir_laptop | Basically the same as with running any software on dd-wrt, I guess |
| 19:25.58 | CunningPike | Hey, dlynes! Long time! |
| 19:25.58 | *** join/#asterisk mosty (n=mosty@213-66-224-163-no22.tbcn.telia.com) |
| 19:25.59 | Spirits-Sight | thanks |
| 19:25.59 | dlynes | Spirits-Sight: Try the nslu2-asterisk mailing list |
| 19:25.59 | tzafrir_laptop | Spirits-Sight, what are your disk-space and memory limitations? |
| 19:25.59 | Spirits-Sight | thank you dlynes |
| 19:25.59 | dlynes | Spirits-Sight: I don't know if they have it working on wrt54g's or not, but they definitely have it working on the Linksys network storage devices |
| 19:25.59 | *** join/#asterisk sprbck (n=sprbck@81.92.203.249) |
| 19:25.59 | Spirits-Sight | flash 8mb and mem is either 16-32mb have to look again |
| 19:25.59 | dlynes | Spirits-Sight: it works on the unslung linux distribution, which is based on debian |
| 19:25.59 | tzafrir_laptop | nslu2 has an extra disk attached to it. This is not the case (by default) with routers |
| 19:26.00 | dlynes | CunningPike: Yeah...living in Brantford, ON now |
| 19:26.00 | dlynes | CunningPike: Moved down here to invest in commercial real estate |
| 19:26.00 | CunningPike | dlynes: Neat - how's it going? |
| 19:26.02 | dlynes | CunningPike: Good...came across one property that we were quite interested in, but got further into it and realized it needed more rehab than we were willing to commit to, so we walked away from it |
| 19:26.02 | dlynes | Looking for another one now |
| 19:26.02 | Badrobot- | hi |
| 19:26.02 | sprbck | Hi. I have a problem with an FXS getting occasionally frozen with a TDM410P. "show channels" shows the port on Rsrvd state, and "zap show channel 1" always shows it as Offhook. I have to stop *, unload modules, reload, start * again to fix it, until next time. Does anyone have any clue about this? |
| 19:26.02 | dlynes | I suspect we'll find another one in the next month or two |
| 19:26.03 | dlynes | CunningPike: if you want, pm me your email address and I can add you to our monthly newsletter to keep you up to date |
| 19:26.21 | CunningPike | dlynes: Should be lots of bargains at, |
| 19:26.21 | CunningPike | *atm |
| 19:26.21 | dlynes | Yeah...but there's a lot of peopel that are wanting me to jeopardize myself to rescue them |
| 19:26.22 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 19:29.22 | dlynes | ~seen flauto |
| 19:29.27 | infobot | flauto <n=zhao@71.194.141.225> was last seen on IRC in channel #asterisk, 513d 1h 52m 29s ago, saying: 'and email'. |
| 19:29.37 | *** join/#asterisk apeiron (n=apeiron@c-76-124-252-61.hsd1.pa.comcast.net) |
| 19:36.00 | *** join/#asterisk deadpigeon (n=deadpige@office.xpressamerica.net) |
| 19:36.55 | deadpigeon | Got a DVG-1402S d-link voip box... long shot but anyone have any success stories with this pile of garbage? |
| 19:37.05 | deadpigeon | 2 days and still can't get the thing to even register with my *. |
| 19:37.20 | _brent_ | i've had lots of bad luck with d-link gear :-( |
| 19:37.42 | deadpigeon | yea.. .this is so frustrating, i've never had much of an issue with their commercial grade routers. |
| 19:38.19 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 19:38.54 | workdraft | <PROTECTED> |
| 19:41.40 | dlynes | deadpigeon: their home office stuff is complete crap....right up there with Linksys's home office stuff |
| 19:41.56 | deadpigeon | right.. |
| 19:41.57 | [TK]D-Fender | Only think i use D-Link for is switches on occasion |
| 19:42.14 | deadpigeon | well, atleast the linksys PAP boxes f***ing work! |
| 19:42.26 | dlynes | I find D-Link actually makes much better quality small office appliances than Linksys |
| 19:42.38 | deadpigeon | damn thing won't even register, I've gone through 6 different firmware versions. |
| 19:42.47 | dlynes | I always have problems with Linksys....DLink I just have problems with most of the time not all of the time |
| 19:42.48 | deadpigeon | never even see attempts in my asterisk debugs. |
| 19:43.21 | deadpigeon | well, apparently they fail completely at making a voip box. never had this much of a problem with the pap2 boxes from linksys |
| 19:43.29 | deadpigeon | granted they suck, but they register and make calls. |
| 19:43.32 | dlynes | deadpigeon: the pap boxes are from some other company that linksys bought |
| 19:43.41 | dlynes | deadpigeon: they're actually originally from sipura |
| 19:43.55 | dlynes | deadpigeon: and yes, they're much more reliable |
| 19:43.59 | deadpigeon | the grandstreams so far have actually turned out to be a better deal. |
| 19:44.44 | *** join/#asterisk saftsack (n=saftsack@p57924807.dip.t-dialin.net) |
| 19:45.40 | dlynes | deadpigeon: have you tried listening on the wire to see if the 1402s is sending anything out? |
| 19:46.05 | dlynes | deadpigeon: i've found at times that the asterisk box doesn't actually show me everything I expect it to |
| 19:46.39 | dlynes | deadpigeon: listening on the wire meaning using a packet sniffer such as iptraf, or ethereal, or ettercap? |
| 19:46.56 | deadpigeon | wireshark. |
| 19:46.59 | dlynes | deadpigeon: ethereal is now called wireshark |
| 19:47.05 | dlynes | deadpigeon: yes, or that one? |
| 19:47.58 | deadpigeon | yea, sadly enough not seeing anything that would indicate why it isn't registering. someone else had suggested that the boxes attempt to auth with clear passwords which is why it wouldnt work for him, but im not even seeing that. |
| 19:48.25 | dlynes | deadpigeon: are you at least seeing traffic in wireshark though? even if it isn't registering successfully? |
| 19:48.40 | deadpigeon | i'm seeing net traffic, no sip traffic |
| 19:49.00 | dlynes | deadpigeon: ok, so no sip traffic, iax2 traffic, or rtp traffic? |
| 19:49.17 | deadpigeon | neither. i dont believe the boxes are iax compatible anyways. |
| 19:49.34 | *** join/#asterisk mweichert (n=mweicher@216.13.154.21) |
| 19:49.48 | mweichert | is a SIP INVITE the same as a SIP registration? |
| 19:49.56 | [TK]D-Fender | mweichert: No. |
| 19:50.08 | [TK]D-Fender | mweichert: an INVITE is an actual call |
| 19:50.34 | [TK]D-Fender | deadpigeon: Enable SIP debug. Not packets means either the device is configured wrong, or you have a firewall / routing issue |
| 19:51.46 | mweichert | [TK]D-Fender, ok, thanks again. I'm still trying to diagnose my dropped call issues. I see the following generated in the log by my phone: EvInviteSuccessResponseTimeout Bye failed - resFE_FAIL?no |
| 19:52.32 | mweichert | if the phone is receiving the call... then would this message indicate that it couldn't send a response to the asterisk server? |
| 19:53.34 | [TK]D-Fender | mweichert: Could be. Go get access to the server. |
| 19:54.13 | mweichert | [TK]D-Fender, ok - I'm still working on that. ISP said that they will have a reply to me tomorrow whether or not they can provide me a log. *sigh* |
| 19:54.43 | mweichert | [TK]D-Fender, thanks again for your help. |
| 19:59.07 | Spirits-Sight | [TK]D-Fender: I was wondering on a small asterisk setup how many ports would be go to use for rft I can make a total of 4 call on my system and I think 4 incoming all at once voicepulse standard connect package |
| 19:59.57 | Spirits-Sight | I read that its not good to have all them ports open at once 10000-20000? |
| 20:01.54 | [TK]D-Fender | spirtYou read wrong, just forward them |
| 20:04.19 | Spirits-Sight | [TK]D-Fender: if I read wrong please explain, it said that the ports range would be open, if that not true then thats fine by me but if its open won't that be a bad thing? |
| 20:04.51 | [TK]D-Fender | Spirits-Sight: Its all the same daemon listening so it isn't really any more secure. Go find something new to neurose about :) |
| 20:07.55 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:09.08 | Spirits-Sight | [TK]D-Fender: I don't care about the ports I just happen to see something that said you should change it to a less of a number, that why I asked. |
| 20:09.45 | *** join/#asterisk setunado (n=fabien@setuns.fr.nf) |
| 20:13.20 | bmoraca | Spirits-Sight: basically, if there is a security hole in the application listening to those ports, it doens't matter if one port or 100,000 ports are open to that application. it takes all of 45 seconds to port scan an IP address, so they'll find them regardless. |
| 20:15.01 | deadpigeon | so, I see the register attempts from wireshark on the new firmware. still doesnt register. |
| 20:15.14 | deadpigeon | couldnt possibly be a nat issue, i've got the thing on a public IP in the same subnet as the asterisk |
| 20:15.20 | deadpigeon | heh. what a pile those dlinks are. |
| 20:15.23 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 20:16.03 | Spirits-Sight | bmoraca: thanks for explaining that to me |
| 20:16.55 | bmoraca | deadpigeon: enable sip debug and pastebin the output |
| 20:18.35 | *** part/#asterisk Spirits-Sight (n=christop@c-76-118-227-40.hsd1.ma.comcast.net) |
| 20:22.28 | *** join/#asterisk werdan7 (i=werdan7@freenode/staff/wikimedia.werdan7) |
| 20:24.18 | *** join/#asterisk Marquel (n=Marquel@port-13761.pppoe.wtnet.de) |
| 20:24.22 | Marquel | morning. |
| 20:26.51 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 20:26.51 | *** join/#asterisk stevetotaro (n=Steve@c-69-142-179-47.hsd1.nj.comcast.net) |
| 20:29.05 | *** join/#asterisk zchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com) |
| 20:29.52 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-da6edb11418826bc) |
| 20:30.48 | Marquel | i have a question about voicmailboxes: is it possible to have one password for all boxes if the mastermailbox is called and also no password at all if a mailbox is called from a certain caller? |
| 20:31.22 | *** part/#asterisk _brent_ (n=_brent_@jive.fttp.xmission.com) |
| 20:32.19 | DavidR2008 | (* 1.4.23.1) I've got a gxp 2020 and I'm using SJphone to test with. If I call the SJphone from the gxp 2020 the SJphone can't hear, if the SJphone initiates the call both ends can hear. Could this be a codec problem? what should I look for on the cli? |
| 20:42.47 | deadpigeon | how can I do a sip debug on a peer that isn't registered? i dont think its going to show me much relative information on an unregistered peer. |
| 20:43.30 | *** join/#asterisk saftsack (n=saftsack@p57924807.dip.t-dialin.net) |
| 20:44.47 | *** join/#asterisk manddy (n=moo@67.204.24.118) |
| 20:48.27 | bmoraca | deadpigeon: just type "sip set debug" |
| 20:48.34 | *** join/#asterisk mandd (n=moo@67.204.24.118) |
| 20:48.34 | bmoraca | you don't have to debug a specific peer |
| 20:49.04 | deadpigeon | right, there's just a lot of debug msgs, sorting through a bunch |
| 20:49.05 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 20:49.15 | bmoraca | there are quite a few, yes |
| 20:49.23 | bmoraca | i'd leave them all there and post the whole bit |
| 20:49.38 | *** part/#asterisk bob_slacker (n=bob_slac@201.47.149.252) |
| 20:50.17 | deadpigeon | right, im gonna try and get the box to re-register while I debug |
| 20:51.34 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-141-35.hrbgpa.dsl-w.verizon.net) |
| 20:54.26 | *** join/#asterisk jcoffi (n=jcoffi@75.147.155.89) |
| 20:55.17 | deadpigeon | bmoraca: http://pastebin.com/m33788f8e |
| 20:55.27 | deadpigeon | i believe i saw the register attempt during that debug |
| 20:56.47 | bmoraca | what's the ip address of the item you're trying to register |
| 20:56.49 | bmoraca | ? |
| 20:57.23 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 20:58.56 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 20:59.12 | telnettech | all....what is the best way to troubleshoot and resolve echo issues with IP phones? |
| 20:59.14 | bmoraca | also, what's the IP address of your server? |
| 20:59.26 | bmoraca | in fact, post your sip.conf as well |
| 21:01.01 | *** join/#asterisk mort___ (n=mort@user-54405220.wfd76b.dsl.pol.co.uk) |
| 21:01.31 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:01.43 | deadpigeon | hang out, i have to find the IP address and I'll give you my sip.conf inna second here. |
| 21:02.33 | *** join/#asterisk nkohh (n=justin@unaffiliated/kohh) |
| 21:03.12 | deadpigeon | 72.8.61.54 = device |
| 21:03.18 | deadpigeon | 72.8.60.21 is asterisk server. |
| 21:03.32 | bmoraca | are either or both behind a NAT? |
| 21:04.08 | nkohh | hello everyone. I'm having a little trouble figuring out how the conditions in "execif" (or if) are supposed to be formed. I'm trying to test against CALLERID(dnid). Would it look something like this? ExecIf($[CALLERID(dnid) == 0000000000]|Whaver|whatever) |
| 21:04.15 | nkohh | formed=formatted |
| 21:05.52 | *** join/#asterisk jpcansa (n=jpbenavi@201.198.231.210) |
| 21:06.16 | deadpigeon | http://pastebin.com/m414f401f <- sip conf for extension |
| 21:06.21 | deadpigeon | at the moment, neither are. |
| 21:06.52 | deadpigeon | I originally had it behind a nat, but I moved it into a public IP hoping it was a nat issue. |
| 21:07.00 | deadpigeon | the server itself is never natted. |
| 21:07.34 | bmoraca | take the nat=yes out of your sip.conf under that peer. also, what's your [general] section look like in sip.conf? |
| 21:07.46 | drmessano | Paste the WHOLE FILE |
| 21:08.14 | bmoraca | that, too |
| 21:09.56 | bmoraca | uhmj |
| 21:10.26 | bmoraca | the user your device is registering as doesn't exist int he snippet of sip.conf you gave me...that's why it's getting a 404 |
| 21:10.29 | deadpigeon | http://pastebin.com/m2ec34c9e <- general. |
| 21:13.06 | deadpigeon | uhm, where are you seeing the user its trying to register at? the username is simply it's extension. its freepbx, the configs are only slightly different as device scripts are auto generated and put in different places. |
| 21:13.25 | bmoraca | Contact: <sip:2073911001@72.8.61.54:5060> |
| 21:13.42 | bmoraca | doesn't look right to me...you've got a 207 prepended on there. |
| 21:14.02 | deadpigeon | hm, yer right.... auth username on the device is set as 3911001 however. =/ |
| 21:14.21 | bmoraca | all i can say is that's not what's being sent to asterisk |
| 21:14.28 | deadpigeon | the device must be taking some other settings from outgoing CID for username. |
| 21:14.32 | deadpigeon | I see that. |
| 21:14.45 | [TK]D-Fender | nkohh: With a valid expression, yes |
| 21:19.32 | *** join/#asterisk nullable_type (n=nullable@hq.verbx.net) |
| 21:21.57 | nullable_type | Hey guys do u know how to use the D option and L option together in DIAL command |
| 21:23.36 | *** join/#asterisk kerx (n=kerx@adsl-69-105-56-225.dsl.irvnca.pacbell.net) |
| 21:23.45 | *** join/#asterisk tecnico (n=tecnico@75.76.169.148) |
| 21:24.02 | *** join/#asterisk _brent_ (n=_brent_@jive.fttp.xmission.com) |
| 21:36.31 | *** part/#asterisk jcoffi (n=jcoffi@75.147.155.89) |
| 21:37.09 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 21:38.32 | *** join/#asterisk AndyCrawford (n=AndyCraw@dynamic-65-161-142-80.tvscable.com) |
| 21:42.55 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:43.06 | ajohnson | Is there any way to have asterisk run a console command immediately after startup? I want the queue log file to be stored using ODBC, but when it initially loads the ODBC connections haven't been registered yet |
| 21:43.17 | ajohnson | Even when you specify preload on the ODBC connections |
| 21:43.37 | ajohnson | But, if you do module reload extconfig after startup, it works fine |
| 21:43.59 | ajohnson | or rather just "reload extconfig" |
| 21:44.12 | drmessano | Sure the ODBC driver is ready? |
| 21:44.23 | drmessano | Maybe you need to delay the start of Asterisk entirely |
| 21:44.43 | *** join/#asterisk columbo (n=PeterFal@pool-173-51-11-230.lsanca.dsl-w.verizon.net) |
| 21:45.49 | ajohnson | Well it appears that one of the 1st few things Asterisk does is load the logger and extconfig |
| 21:46.30 | ajohnson | so because Asterisk hasn't read the odbc config file yet, the odbc engine isn't registered yet |
| 21:46.35 | drmessano | I dont mean the Asterisk ODBC connector |
| 21:46.40 | ajohnson | odbc sip peers and everything else works just fine |
| 21:46.44 | drmessano | ok |
| 21:47.15 | ajohnson | It is specific to queue_log and the fact that it is written to use the logger, not the sip modules, etc |
| 21:47.27 | ajohnson | and logger is one of the 1st things to load, and you can't change that |
| 21:47.39 | ajohnson | so I need to run a command immediately after loading to reload extconfig |
| 21:49.24 | drmessano | I wasnt suggesting changing when logger loads |
| 21:49.55 | drmessano | I was suggesting delaying asterisk's startup entirely |
| 21:51.08 | ajohnson | How would that help? |
| 21:51.43 | drmessano | If you stop and start asterisk right now, will ODBC start up correctly? |
| 21:51.47 | drmessano | JUST asterisk |
| 21:51.54 | ajohnson | I see what you're saying |
| 21:52.01 | ajohnson | ODBC starts just fine |
| 21:52.10 | ajohnson | And I'm stoping and restarting asterisk after system startup |
| 21:52.14 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 21:52.24 | ajohnson | It isn't a boot order thing, it is an asterisk load order thing |
| 21:53.23 | drmessano | So if you stop and restart asterisk right now, you will still have to reload? |
| 21:53.43 | _brent_ | ajohnson: what are you preloading in modules.conf? |
| 21:54.11 | ajohnson | preload => res_odbc.so |
| 21:54.11 | ajohnson | preload => res_config_odbc.so |
| 21:54.32 | ajohnson | Except modules.conf explicitly says preload happens after logger initialization |
| 21:54.38 | ajohnson | which means it is already too late in this case |
| 21:55.03 | *** join/#asterisk _pepo_ (n=pepo@200.55.224.2) |
| 21:55.06 | _brent_ | well, that sounds like a chicken and egg problem |
| 21:55.09 | _pepo_ | hi friends |
| 21:55.26 | _brent_ | i'm sure the devs intended to have the logger load first to troubleshoot odbc/etc problems |
| 21:55.35 | _brent_ | but you want to log to odbc |
| 21:55.38 | _brent_ | right? |
| 21:55.50 | _pepo_ | is there some way to use sipp to test voicemail? (I mean use sound files to record in voicemails) |
| 21:55.54 | ajohnson | Yeah |
| 21:56.00 | ajohnson | essentially |
| 21:56.17 | ajohnson | the way the queue_log realtime patch was written, it uses the logger (duh) |
| 21:56.41 | ajohnson | so ultimately what would work best would be to reload extconfig after Asterisk has completed loading |
| 21:56.53 | *** join/#asterisk orkid (n=orkid@unaffiliated/orkid) |
| 21:57.02 | ajohnson | which reloads sip, queues, logger, etc |
| 21:57.14 | _brent_ | right...hmmm |
| 21:57.25 | ajohnson | There is a way to tell Asterisk to run a script as a source of a config file |
| 21:57.28 | ajohnson | I forget how to do that |
| 21:57.35 | _brent_ | #exec or something |
| 21:57.38 | ajohnson | Otherwise, I would have extensions.conf run a script |
| 21:57.39 | ajohnson | that's it |
| 21:58.00 | ajohnson | so when it got around to processing extensions.conf, those modules would already be loaded, I can reload them |
| 21:58.04 | ajohnson | we can try :) |
| 21:58.08 | _brent_ | :-) |
| 21:58.24 | Mark21 | is there somewhere an example for how to create the following (please note, no gui or web interface, just the asterisk configs are available): every hour someone is called and will here some moh (or something similair) that is saying press 1, after he pressed 1 the call will be ended. if he doesn't answer the phone within 1 minute and/or doesn't press 1 within 30 sec. after answering the phone > call a second person with the same system (read picku |
| 21:58.35 | _brent_ | another option would be to run something in your init script (assuming you're starting * that way) |
| 21:59.09 | ajohnson | Hmmm |
| 21:59.11 | ajohnson | that would work |
| 21:59.15 | orkid | I'm trying to do an extensions match by callerid. I have an extension called _1NXXNXXXXXX/spa_fxs. When I call a valid number from spa_fxs it says Call from 'spa_fxs' to extension '12127483922' rejected because extension not found. Why can't it match the extension since it knows it's calling from 'spa_fxs' ? |
| 21:59.21 | ajohnson | would have to tell the init script to sleep for a couple seconds |
| 21:59.51 | ajohnson | because spa_fxs is not the caller ID |
| 21:59.54 | ajohnson | It is the device |
| 22:00.10 | orkid | How can I see what the caller id is? |
| 22:00.17 | orkid | there is no 'Get' equivalent of 'Set' |
| 22:00.26 | orkid | or is there an 'Echo' to console command? |
| 22:00.28 | ajohnson | is this a SIP device? |
| 22:00.40 | orkid | yes. It's an SPA-3102 (the fxs sip client portion) |
| 22:00.48 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-2a303c9b4dd65a56) |
| 22:01.06 | ajohnson | so why do you want to match on that device specifically? |
| 22:01.20 | ajohnson | you can just change the context= section of sip.conf to make it use a different context for dialing |
| 22:01.28 | orkid | yes. I want to math that device (I don't know what it's callerid is I suppose) |
| 22:01.46 | orkid | Is there a way to check its caller-id value? |
| 22:02.18 | orkid | It seems cleaner that way, with less duplication |
| 22:02.51 | _brent_ | orkid: look for the ex-girlfriend screening sample |
| 22:03.07 | _brent_ | but i agree with ajohnson about using a different context for it |
| 22:03.10 | orkid | yup, I did. I guess I'll try "NoOp(${CALLERID}) |
| 22:03.15 | orkid | to see what the CID is |
| 22:03.32 | orkid | that's what i needed to find, a cmd that will just echo to console |
| 22:03.33 | ajohnson | right |
| 22:04.05 | ajohnson | You're trying to use Asterisk in a way that is unintended. I think you're heading down a difficult path and you're going to be fighting Asterisk the whole way |
| 22:04.24 | ajohnson | the context setting in sip.conf is there to do exactly what you're asking |
| 22:05.00 | orkid | it doesn't echo anything. why not? |
| 22:05.09 | ajohnson | echo? |
| 22:05.16 | ajohnson | NoOp will print lines to the screen |
| 22:05.28 | orkid | .. print CID to console. Will it not print anything if there is no CID set? |
| 22:05.30 | ajohnson | But obviously it isn't outputting anything because it isn't matching on anything |
| 22:05.51 | orkid | ... re context change: I would have to duplicate the whole extensions table simply for this. |
| 22:06.08 | ajohnson | Then you have a deeper design problem |
| 22:06.10 | orkid | I changed it to match on all extensions (took out /spa_fxs), so it matches.. |
| 22:06.22 | orkid | just doesn't give me anything other than 'spawn extension ...' |
| 22:06.28 | ajohnson | Can you pastebin the console output? |
| 22:06.37 | Mark21 | is there a way asterisk will dial a certain phone number on a certain trunk and call on the other end a certain IVR (so the person that picks up the phone will hear the IVR playing, requesting to press 1) |
| 22:06.41 | *** join/#asterisk desdesdesdes (n=kg@196.211.34.2) |
| 22:07.09 | desdesdesdes | can someone please explain rxgain and txgain? |
| 22:07.10 | ajohnson | Mark21: Yes, though I don't know how :-D |
| 22:07.16 | orkid | just three lines: |
| 22:07.16 | orkid | <PROTECTED> |
| 22:07.16 | orkid | <PROTECTED> |
| 22:07.20 | orkid | <PROTECTED> |
| 22:07.43 | ajohnson | Did you set callerid in sip.conf? |
| 22:07.58 | orkid | callerid=spa_fxs is there |
| 22:08.13 | orkid | in sip.conf. hm |
| 22:08.41 | orkid | or maybe should be <spa_fxs> |
| 22:08.57 | ajohnson | slow down and read :) |
| 22:09.02 | Mark21 | ajohnson: i did know it is possible, i just need to know the name so i can find it |
| 22:09.07 | [TK]D-Fender | orkid>yup, I did. I guess I'll try "NoOp(${CALLERID}) <- and pick something VALID |
| 22:09.12 | ajohnson | <> denotes the numberic component of clid |
| 22:09.39 | orkid | sigh. i thought ${CALLERID} would give it all |
| 22:09.45 | ajohnson | ${CALLERID(name)} |
| 22:09.49 | ajohnson | No... |
| 22:10.49 | orkid | "Anonymous" <spa_fxs> |
| 22:11.04 | orkid | Do you know of a way to set this not in sip.conf but in the spa3102 config pages? |
| 22:11.23 | orkid | I couldn't figure it out. I set Display Name but it does not get sent as CID Name it seems |
| 22:11.49 | _brent_ | orkid: have you tried dropping the spa into a different context? it seems like that would be simple |
| 22:12.09 | ajohnson | He doesn't want to |
| 22:12.15 | orkid | yes ajohnson mentioned that, but I would have to copy/paste all of the extensions right? |
| 22:12.21 | _brent_ | no |
| 22:12.31 | ajohnson | you can use the include => directive |
| 22:12.39 | _brent_ | just go to the other context, do your special stuff and then ... |
| 22:12.40 | _brent_ | exactly |
| 22:12.44 | ajohnson | to include another ocntext |
| 22:13.04 | _brent_ | or just go to the other context |
| 22:13.37 | orkid | ok. so say I include [outbound] context... and now I dial '1234' from spa_fxs. It goes to its own context... How would I run priorities in that context before it going to the inluded [outbound] context? |
| 22:14.07 | _brent_ | just put your normal exten => whatever lines in there |
| 22:14.07 | ajohnson | Orkid, what are you really trying to DO |
| 22:14.09 | orkid | I can just have an exten => 1234,1,whatever in [spa_fxs] ... but then once it's done processing it'll just stop |
| 22:14.17 | ajohnson | Not how are you trying to do it, what are you trying to do |
| 22:14.21 | orkid | I'm trying to set outbound CID based on the extension placing the call. |
| 22:14.32 | ajohnson | Ok |
| 22:14.57 | ajohnson | So SIP/ajohnson makes a call to 6025551234 |
| 22:15.03 | ajohnson | You want to change my CLID on my call? |
| 22:15.07 | orkid | yes |
| 22:15.19 | _brent_ | [spa_outbound] |
| 22:15.19 | _brent_ | exten => _X.,1,DoSomething() |
| 22:15.19 | _brent_ | exten => _X.,2,Goto(outbound|${EXTEN}) |
| 22:15.30 | orkid | oh. ok :) |
| 22:15.38 | ajohnson | I'm glad we had this conversation because you're going about this the wrong way :) |
| 22:16.12 | _brent_ | http://www.voip-info.org/wiki-Asterisk+cmd+goto |
| 22:16.17 | ajohnson | 1st of all, you can set callerid in sip.conf for the peers(extensions) you are worried about |
| 22:16.23 | orkid | really? It seems simpler to me to do it that way.. But I guess you're right. This keeps client specific things in their own context before doing the actual diailing (in the dialing context). correct? |
| 22:16.45 | ajohnson | 2nd of all, there is a setvar directive in sip.conf that can set variables on calls |
| 22:16.58 | orkid | ajohnson: I learned about this after I started working on getting this to work, and wanted to figure out why it wasn't working to begin with :) |
| 22:17.05 | orkid | but yes |
| 22:17.17 | ajohnson | right |
| 22:17.18 | ajohnson | so |
| 22:17.30 | orkid | So the best way I guess is to just set callerid= in sip.conf and that's it, right? |
| 22:17.40 | ajohnson | in sip.conf for ajohnson you do setvar=externclid=6025551234 |
| 22:17.57 | ajohnson | then in your outbound context you can use a line that goes something like this |
| 22:18.01 | orkid | why not callerid=<6025551234> ? |
| 22:18.27 | ajohnson | exten => _NXXNXXXXXX,1,Set(CALLERID(num)=${externclid}) |
| 22:18.43 | ajohnson | that works too, but then it will apply to all calls |
| 22:18.46 | ajohnson | including internal calls |
| 22:19.02 | orkid | Oh, yes. |
| 22:19.07 | ajohnson | you can set callerid to an extension number for internal calls and then use the externclid variable for external calls |
| 22:19.26 | orkid | That makes more sense yes :) |
| 22:19.37 | orkid | You've been here, done this, before? |
| 22:19.39 | orkid | :) |
| 22:20.04 | ajohnson | Heh, been doing it a while |
| 22:22.15 | *** join/#asterisk knarfly (n=vtserije@c-75-74-113-9.hsd1.fl.comcast.net) |
| 22:22.22 | knarfly | hey gonza |
| 22:22.26 | knarfly | hey gonzo |
| 22:25.49 | knarfly | just reinstalled the OS on my FreeBSD server and the installed Asterisk-1.4.23.1 and Zaptel-1.4.11. And zttool appears to show my X101P card is not there. I can put the old hard drive back in and everything works. Is there something new with Zaptel on FreeBSD ? |
| 22:29.34 | Mark21 | ajohnson: http://www.voip-info.org/wiki-Asterisk+auto-dial+out << how to just dial an extencion and if succesfull do nothing (or I would need an other empty extencion to let it call that and that thing does nothing), because i missed the option at http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out+deliver+message to specify the time for the call |
| 22:30.00 | ajohnson | Cool |
| 22:37.48 | orkid | Hm, ok. so far so good. However, What I do is call through another server. So server1 calls 16025551234@server2. Now server2 shows "Unknown" <Unknown> for CALLERID(all). There is no callerid= set for server1 in sip.conf of server2. |
| 22:38.03 | orkid | server1 registers to server2 |
| 22:40.09 | ajohnson | CLID shows up fine on server 1? |
| 22:40.14 | orkid | yes |
| 22:41.06 | orkid | basically I'm calling SIP/16025551234@server2 |
| 22:41.18 | orkid | Dial(...) |
| 22:42.24 | ajohnson | _brent_: BTW, putting exec in extensions.conf fixes the issue :) |
| 22:42.32 | _brent_ | nice |
| 22:42.43 | ajohnson | the script reloads extconfig, which reloads logger |
| 22:43.01 | ajohnson | now queue log messages go straight to an odbc connection :) |
| 22:43.51 | ajohnson | orkid: So you're setting CLID before your dial command? |
| 22:44.40 | ajohnson | Set(${CALLERID(num)}=${externclid}) ? |
| 22:44.44 | orkid | I'm setting in on server1. and then I'm doing a NoOp CALLERID(all) to see that's set. (Anonymous <12125551234>) Then I Dial(16025551234@server2) |
| 22:45.15 | orkid | I do a NoOp CALLLERID(all) on server2 (in extension _1NXXNXXXXXX) but see "Unknown" <Unknown> |
| 22:45.28 | ajohnson | Weird |
| 22:45.53 | orkid | They're different asterisk versions. server2 is 1.6, server1 is 1.4 |
| 22:46.07 | ajohnson | You shouldn't have to do this, but it is possible to pass information between servers using SIPAddHeader |
| 22:46.13 | *** part/#asterisk _brent_ (n=_brent_@jive.fttp.xmission.com) |
| 22:46.18 | ajohnson | and SIPGetHeader |
| 22:46.30 | ajohnson | pass the CLID as a variable in the header and then set it on the other side |
| 22:46.37 | ajohnson | but you definitely shouldn't have to do that |
| 22:46.57 | orkid | Are version differences relevant? |
| 22:47.02 | ajohnson | always |
| 22:47.14 | ajohnson | You never know what's going to be broken :) |
| 22:47.25 | orkid | 1.6 (server2) and 1.4 (server1) even? |
| 22:47.36 | ajohnson | I don't know, haven't worked with that configuration yet |
| 22:47.59 | ajohnson | I would look for something in CHANGES or sip.conf example on the 1.6 server |
| 22:48.07 | ajohnson | relating to clid |
| 22:48.11 | ajohnson | or sip |
| 22:49.12 | orkid | Maybe I'll try 1.4 on server2 |
| 22:49.33 | ajohnson | Good luck, I'm off for the day |
| 22:49.41 | orkid | thanks for your help |
| 22:53.54 | *** join/#asterisk moy (n=chatzill@74.12.124.89) |
| 22:54.23 | *** join/#asterisk `paul (n=kutimoy@121.97.99.151) |
| 22:54.51 | *** join/#asterisk timeshell (n=chatzill@206.248.136.108) |
| 23:02.26 | *** join/#asterisk jcoffi1 (n=jcoffi@208.87.0.146) |
| 23:03.22 | orkid | hmm, now 1.4.17 (ubuntu) -> 1.4.24 (vanilla) and still showing Unknown <Unknown> |
| 23:05.18 | *** join/#asterisk jcoffi (n=jcoffi@75.147.155.89) |
| 23:08.28 | *** part/#asterisk jcoffi (n=jcoffi@75.147.155.89) |
| 23:09.00 | *** join/#asterisk path_ (n=path_@71-119-21-190.adsl.terra.cl) |
| 23:09.39 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-194-244.lns10.mel6.internode.on.net) |
| 23:09.50 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 23:10.49 | *** join/#asterisk RobertLaptop (n=rmiddle@75.145.84.181) |
| 23:12.23 | *** join/#asterisk Vec (n=Vec@196-210-172-92-rrba-esr-3.dynamic.isadsl.co.za) |
| 23:14.45 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 23:22.21 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 23:44.29 | *** join/#asterisk JuStIcIa_ (i=john@cbl-sd-74-96.aster.com.do) |
| 23:46.00 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 23:57.33 | *** join/#asterisk mazpe (n=mazpe@c-71-196-32-22.hsd1.fl.comcast.net) |
