| 00:00.45 | jaytee | time for some din-din, bbiab |
| 00:02.01 | sprite-- | jaytee: Let me know when you get back. Ran into another issue now. If I answer on my softphone it keeps rining on my cell, it fails to transmit the answer packet. |
| 00:02.15 | sprite-- | [Dec 3 19:01:04] WARNING[17598]: chan_sip.c:2819 retrans_pkt: Hanging up call 33826469-3437337634-425521@msw2.telegeny.net - no reply to our critical packet (see doc/sip-retransmit.txt). |
| 00:06.02 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 00:06.43 | sprite-- | Retransmitting #3 (no NAT) to 204.11.192.37:5060: |
| 00:06.48 | sprite-- | Shouldn't that be showing NAT? |
| 00:08.04 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 00:08.46 | drmessano | Is there a valid view= string in SVN URL |
| 00:08.54 | drmessano | Is there a valid view= string in SVN URL's to show plain text |
| 00:09.16 | drmessano | Trying to view the changelog for 1.6.0.3-rc1 and firefox keeps locking up |
| 00:18.33 | japerry | heya folks, anyone in here using freepbx? |
| 00:18.46 | japerry | http://www.freepbx.org/trac/wiki/DevicesTakeTwo --> looks really old, I'm trying to find a neat way to provision phones |
| 00:19.30 | jblack | japerry: Do you normally walk into a ford dealership and ask which salesmen own a honda? |
| 00:19.56 | japerry | jblack: freepbx is asterisk and digium...? |
| 00:20.32 | jblack | If freepbx is asterisk, then why do they call it freepbx? |
| 00:21.09 | Carlos_PHX | Mazda makes many of Ford's vehicles, but you still go to the Ford dealership for parts for your Mazda-made Ford. |
| 00:21.17 | japerry | nm. I guess they aren't close enough |
| 00:21.28 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-64c0ca5622bf1b97) |
| 00:21.34 | jblack | No, they're not. Good call. :) |
| 00:21.36 | Carlos_PHX | In any case, I've never seen anyone here who admits to having used one of the GUI versions for any length of time. |
| 00:22.07 | drmessano | japerry: I think what jblack is trying to say, is try #freepbx |
| 00:22.27 | Corydon76-dig | A GUI is great if you need a minimal interface to do minimal changes |
| 00:22.27 | japerry | drmessano: yes I'm in there ;-) thanks though. I didn't realize there was such a gap between the two projects |
| 00:22.59 | drmessano | japerry: No, indignant GUI hatred is more like it |
| 00:23.03 | jblack | There's a small code gap, and a huge political one, between the two. |
| 00:23.06 | japerry | drmessano: apparently |
| 00:23.45 | Corydon76-dig | drmessano: Name one GUI that gives you all the power to change things that people playing with config files can change |
| 00:23.52 | japerry | too bad too. beccause I'm not going to have my secretary looking at the asterisk CLI to see who is on the line |
| 00:23.54 | jblack | It's a hatfield/mccoy sort of thing. |
| 00:24.15 | Corydon76-dig | jblack: No, it's really not |
| 00:24.18 | Carlos_PHX | I've tried many times to use a GUI but there's always one thing it can't do that I want. |
| 00:24.23 | Carlos_PHX | Asterisk is like crack. |
| 00:24.36 | Corydon76-dig | jblack: every GUI gives you a subset of features that you can configure |
| 00:24.37 | drmessano | Corydon76-dig: Not sure why that is even relevant.. name one Linux desktop that lets you change every OS option point and click... |
| 00:24.50 | Corydon76-dig | drmessano: exactly |
| 00:25.00 | jblack | Corydon76-dig: Yes, it really is. Doesn't mean either you or I are on one side or another, but there's definitely sides. |
| 00:25.08 | drmessano | Corydon76-dig: Ok, so you made my point? |
| 00:25.10 | Corydon76-dig | drmessano: A GUI by its very nature does not give you complete control over a system |
| 00:25.24 | japerry | yes folks, because clearly point-n-click is not the future. sigh |
| 00:25.33 | drmessano | Corydon76-dig: Ok, I dont think they intent to do that |
| 00:25.38 | drmessano | intend* |
| 00:25.50 | japerry | Corydon76-dig: I call BS. poorly designed GUIs don't give you complete control over a system. |
| 00:25.59 | Corydon76-dig | drmessano: that's why many people here eschew the very limited GUI because they need to do MORE |
| 00:26.16 | Corydon76-dig | japerry: Name a well-designed GUI |
| 00:26.29 | japerry | for asterisk? none |
| 00:26.35 | Corydon76-dig | japerry: For anything |
| 00:26.55 | Corydon76-dig | japerry: and I bet I can find a setting that you can change with a config file that you can't change with the GUI |
| 00:27.06 | drmessano | Corydon76-dig: It's more than "eschew".. but anyway.. Not getting into this again. |
| 00:27.22 | coppice | no GUI ever gives complete control, and I guess that's inevitable. what is bad is that most GUIs massively conflict with any manual changes you make. that's avoidable in a good design |
| 00:27.46 | japerry | Gallery2, drupal, most well designed web apps |
| 00:27.52 | japerry | Yast2 comes close... |
| 00:28.02 | japerry | (but that doesn't meet your specification) |
| 00:28.37 | Corydon76-dig | japerry: all web apps require command line setup before they will run |
| 00:28.37 | japerry | my point is that GUIs are not inherently bad, and are necessary for the 'folks downstairs' to be able to efficiently use a tool |
| 00:29.06 | Corydon76-dig | japerry: I never said I had a problem with the people who are not in engineering using GUIs |
| 00:29.09 | japerry | Corydon76-dig: not true. unzip the drupal app on your desktop in windows, drag-n-drop into an ftp client, goto your webbrowser |
| 00:29.21 | Corydon76-dig | japerry: but the majority of us in here are in Engineering and IT and NEED that control |
| 00:29.34 | Corydon76-dig | japerry: I don't use Windows |
| 00:29.50 | japerry | s/Windows/OSX/g or s/Windows/Linux/g |
| 00:30.02 | Corydon76-dig | japerry: I really doubt that |
| 00:30.04 | jblack | There's some things I wouldn't want to do without a gui. Playing chess, for example. There's other things I wouldn't want to do with a gui.. configuring a phone system is an example. I don't think there's a rational way to visually configure something that is both flexible and complicated. |
| 00:30.13 | japerry | Corydon76-dig: I don't use windows either :-P |
| 00:30.44 | Corydon76-dig | japerry: but even in my Linux GUI, there are things I have to configure with a file |
| 00:31.23 | Corydon76-dig | I can either abide by a designer's choice not to include an element or I can fiddle with an option and get the behavior I want |
| 00:31.56 | Corydon76-dig | I, myself, have written GUIs for editing content and add/change/delete Asterisk users |
| 00:31.58 | japerry | true, but I'm not concerned as much about the configuration being GUI as I am the end user administration. I have some tier-1 IT people who need the ability to easily provision phones without needing to get into CLI |
| 00:32.31 | Corydon76-dig | But just because I expect some people to use my GUI, I'm not going to use one for anything but the most mundane tasks |
| 00:32.41 | japerry | I want to offload as much of the end configuration as possible, so every time we get a new employee, or desk move, that I don't have to be the one to do it. so I can work on more important things |
| 00:33.02 | drmessano | japerry: Naw, they need to learn how to hand edit asterisk configs too.. |
| 00:33.04 | Corydon76-dig | japerry: precisely. But for the control you want, you're going to have to write that GUI |
| 00:33.18 | jblack | So make a gui yourself that allows for modifications that make sense to them. That's what I do for my clients. |
| 00:33.23 | Corydon76-dig | japerry: but you're also going to write a specialized dialplan |
| 00:34.05 | Corydon76-dig | This channel isn't as much for the people who just want a phone system to work as it is for the IT people who need to configure that system and specialize it to their business |
| 00:34.35 | Corydon76-dig | and that's why most people in here aren't using a common GUI set |
| 00:34.48 | jblack | Apache and a few php pages, and they can add/remove voicemail, adjust who sits at what extension. Making such a thing is part of your job. Not the job of the tool itself. :) |
| 00:34.48 | coppice | the average person would find config files much easier to work with if they were in Chinese |
| 00:34.49 | japerry | well I'm in the right channel then :-P my requirements have a GUI bit, but I won't ask about that here |
| 00:35.19 | jblack | coppice: Hmmm. Intersting point. |
| 00:35.30 | drmessano | waits for the "if you even have a GUI in the same house, GTFO" bit.. |
| 00:36.11 | japerry | jblack: I'm more interested in engineering backend stuff like LDAP integration, converging communication mediums, etc... and offloading the easy day-to-day admin to those who work helpdesk |
| 00:36.54 | jblack | Yes. And what's just what I'm saying. |
| 00:36.58 | japerry | because most helpdesk people I've met aren't that smart, and mess up config files if you let them have raw access.. IE phpMyAdmin or even raw mysql CLI... I want to control what they do |
| 00:37.09 | jblack | You pick the tool, you design the interface, and let the monkeys do the day to day work. |
| 00:37.31 | japerry | and I was hoping that most of the interface is already done, so I can move on to the next thing |
| 00:37.47 | japerry | I don't like duplicating work if something already exists out there and can be efficently used |
| 00:37.52 | jblack | You won't find such a thing, for aforementioned reasons. |
| 00:38.32 | jblack | Some use ldap, some don't. Some use an sql server, some don't. Some want a web based fop, some don't. Some want balancing between servers, others barely have one. |
| 00:38.39 | japerry | well and thats why I came in here. someone posted work on a autoprovisioning tool within freepbx, so I wanted to know if anyone has worked with it. which this is obviously the wrong channel to ask this in :-P |
| 00:39.20 | Corydon76-dig | japerry: there's an autoprovisioning tool in 1.6, called res_provision |
| 00:39.21 | jblack | Look at combining dhcp and apache. Set something up so that if someone types in the mac for a phone, it sets up the provisioning for them.] |
| 00:39.48 | Corydon76-dig | errr, res_phoneprov, rather |
| 00:39.58 | japerry | Corydon76-dig: Thanks! =D |
| 00:40.09 | Corydon76-dig | but it's for provisioning the phone settings, not for provisioning users |
| 00:40.21 | japerry | Corydon76-dig: thats exactly what I'm looking for.. phone settings |
| 00:40.45 | japerry | the user provisioning seems to work okay in Freepbx so far.. I'm looking to get LDAP integration so I can just set an LDAP bit to assign a user to an extension |
| 00:40.52 | [TK]D-Fender | Corydon76-dig: i DON'T REALY SEE WHY EFFORT WAS MADE FOR A RES LIKE THAT IN THE FIRST PLACE... DOESN'T REALLY BELONG |
| 00:40.55 | [TK]D-Fender | darn caps.. |
| 00:41.37 | Corydon76-dig | [TK]D-Fender: There was a business need for it by an independent customer, and rather than keeping the tool proprietary, it made its way into open source |
| 00:41.50 | jblack | [TK]D-Fender: Sayin' it like you mean it on accident? :) |
| 00:42.26 | sprite-- | http://rafb.net/p/mhadml39.html any ideas why it's not working? |
| 00:42.43 | Corydon76-dig | People do pay for development, you know |
| 00:43.26 | japerry | Corydon76-dig: is the res_phoneprov a standard addon for 1.6 |
| 00:43.38 | scooby2 | whats the diff between 'exten => 6001,1,queue(6001|t)' and 'exten => 6001,1,Queue(${EXTEN}|t|||60)' |
| 00:43.39 | Corydon76-dig | japerry: not an addon... it's in the core |
| 00:43.45 | japerry | Corydon76-dig: oh awesome |
| 00:44.01 | japerry | so just place the phoneprov.conf in the /etc/asterisk folder, configure it and it should work? ;-) |
| 00:44.19 | Corydon76-dig | scooby2: what's the difference between 60 seconds and an eternity? |
| 00:44.36 | Corydon76-dig | japerry: I'm not quite that familiar with it |
| 00:44.48 | japerry | Corydon76-dig: tis okay. I think I found what I need. thanks a bunch |
| 00:44.53 | [TK]D-Fender | Corydon76-dig: Yeah I understand that its a "customer paid" handover to the community... but you give an inch and users want to take a mile... |
| 00:45.13 | [TK]D-Fender | Corydon76-dig: Just afraid it'll blossom with every schmucks expectations... |
| 00:45.32 | [TK]D-Fender | Corydon76-dig: Which of course for me is just a distraction for the real stuff everone really wants. |
| 00:45.37 | Corydon76-dig | Expectations and $3 will buy you a cup of coffee |
| 00:46.12 | Corydon76-dig | Expectations don't pay our bills, nor do they get code into the source |
| 00:46.16 | drmessano | If everyone suddenly wants it to blossom, how is it not what everyone wants? |
| 00:46.22 | drmessano | Sounds a bit self serving to me.. |
| 00:46.33 | sprite-- | [TK]D-Fender: I posted full SIP logs. Any clue why my connection is not working? Everything worked fine with version 1.4, I'm having no luck with version 1.6 |
| 00:46.41 | scooby2 | Corydon76-dig: sorry i knew that part. Does the case matter for queue and ${EXTEN}? |
| 00:47.00 | scooby2 | ${EXTEN} versus hardcoded |
| 00:47.21 | Corydon76-dig | ${EXTEN} is case-sensitive. QuEuE is not |
| 00:47.32 | *** join/#asterisk el_critter (n=newbie@190.142.70.148) |
| 00:47.39 | el_critter | hi |
| 00:47.58 | *** join/#asterisk jer_ (n=jtregunn@unaffiliated/jer) |
| 00:48.24 | scooby2 | Corydon76-dig: thank you |
| 00:49.35 | el_critter | I can´t dial a valid dialplan etxension when I pickup the headset of my Polycom but I can do it when I dial first and then pickup the headset. Can you explain my why? |
| 00:49.45 | seanbright | digit map |
| 00:49.59 | seanbright | that'll be $125 |
| 00:50.25 | el_critter | paypal? :P |
| 00:50.36 | seanbright | heh |
| 00:50.53 | el_critter | thx |
| 00:50.55 | seanbright | yup |
| 00:52.20 | [TK]D-Fender | seanbright: rip-off.... for that much I would have built the provisioning server and done his configs & templates for him :p |
| 00:52.35 | [TK]D-Fender | sprite--: pastebin your sip.conf masking only passwords |
| 00:53.48 | sprite-- | http://rafb.net/p/U2j8bb37.html |
| 00:54.42 | [TK]D-Fender | sprite--: namy settings wrong : read up |
| 00:54.43 | [TK]D-Fender | ~sipnat |
| 00:54.44 | 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 |
| 00:54.46 | [TK]D-Fender | many* |
| 00:54.49 | [TK]D-Fender | ^^^^^ |
| 00:55.47 | beek | Good evening all. I'm having issues with * 1.6 and IMAP storage of voicemail. I can set a temporary greeting but when I tell it to be deleted, I get the message that it has but the files remain. In the app_voicemail.c source there's a bunch of "ast_debug" statements. What does one need to do to get those activated so that I can see what * is attempting to do? |
| 00:56.48 | beek | core set debug atleast 'n' doesn't seem to be the ticket. |
| 00:57.11 | seanbright | beek: have you configured logger.conf to include debug info? |
| 00:57.45 | beek | seanbright: Crap... that would be it. Thanks! |
| 00:57.52 | seanbright | that'll be $125 |
| 00:58.11 | seanbright | i am such a whore. |
| 00:58.14 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 00:58.16 | beek | I'll have to make payment arrangements! |
| 01:01.31 | sprite-- | [TK]D-Fender: I fixed the allowreinvite to canreinvite and added nat=yes under the callcentric context. Did you spot any other errors? because it's still not working and these settings worked fine in 1.4 |
| 01:01.37 | beek | Well, that got me lots of output but not enough information to track this down... |
| 01:01.47 | seanbright | beek: what release of 1.6? |
| 01:01.53 | beek | 1.6.0.1 |
| 01:02.21 | seanbright | it might already be resolved in the branch, lemme look-see |
| 01:02.58 | beek | seanbright: Funny thing is that it's also sorting message 0 on my IMAP server with the temporary message. |
| 01:03.35 | beek | s/sorting/storing/ |
| 01:03.47 | beek | (I love jbot) |
| 01:05.24 | beek | waits with bated breath hoping for an affirmative answer... |
| 01:05.27 | *** join/#asterisk chendy (n=chatzill@58.60.30.239) |
| 01:05.54 | seanbright | hmmm |
| 01:06.02 | seanbright | haven't seen anything IMAP related |
| 01:06.09 | sprite-- | [TK]D-Fender: I am tired of trying to figure this out. Help me figure it out by PM and if we can find a solution within the next 30 mins I'll paypal you $100. |
| 01:07.24 | seanbright | beek: and i see nothing on bugs.digium.com related to that specific problem |
| 01:07.34 | beek | seanbright: I even used "imapgreetings=no" in an attempt to get asterisk to store the greetings on the system, not in the IMAP server. |
| 01:07.35 | japerry | so here is a fun non-gui question... anyone try to make asterisk do LDAP lookups for user authentication? |
| 01:07.56 | beek | seanbright: I could try posting first to the mailing list and if no joy there file a bug report. |
| 01:08.01 | scooby2 | neat, I figured out what is causing my issue I believe. Now to just try and figure out why it is happening. Calls come into my [queues] section of extensions.conf. internal phones use the [local] section. For some reason when trying to transfer a call to a different queue, users in [local] are sending the calls to the number in [queues] and not in [local]. That make any sense? |
| 01:08.29 | seanbright | beek: hmmmm |
| 01:08.30 | beek | I looked at bugzilla for 1.6 series and only saw three items pertaining to IMAP. |
| 01:08.31 | seanbright | beek: http://bugs.digium.com/view.php?id=13414 |
| 01:08.32 | scooby2 | queues is not included anywhere |
| 01:09.04 | scooby2 | do i need to specify which context is used when transfering calls? |
| 01:09.49 | beek | seanbright: Perhaps related... I haven't tried busy/unavailable yet. |
| 01:10.27 | seanbright | that rev was blocked from trunk |
| 01:10.35 | seanbright | so i assume it wasn't applicable |
| 01:11.01 | el_critter | thx again, bye! |
| 01:11.05 | beek | seanbright: I have this * box between the legacy PBX and telco, so * can't check to see if the extension is busy. Temporary greeting would work great for this purpose as a "vacation message". |
| 01:11.53 | seanbright | beek: try the mailing list first, then a bug report. |
| 01:12.16 | seanbright | beek: the IMAP and ODBC storage mechanisms don't get enough of a workout in the wild i'm afraid |
| 01:12.45 | seanbright | and that code frightens me |
| 01:13.26 | beek | seanbright: It is scary. I haven't done C in years but IMAP storage was one of the selling points for adding this * box, so I'm willing to trace this a bit. |
| 01:14.02 | *** join/#asterisk ZaVoid (n=zavoid@nj-76-6-39-193.dhcp.embarqhsd.net) |
| 01:14.03 | seanbright | beek: excellent. |
| 01:14.09 | beek | seanbright: Thanks for your input and I'll follow up through the mailing list. |
| 01:14.14 | seanbright | sounds good |
| 01:14.16 | seanbright | good luck! |
| 01:17.18 | *** join/#asterisk dssman (n=no@CPE001d7e602900-CM0011aec52a9c.cpe.net.cable.rogers.com) |
| 01:18.51 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 01:18.56 | scooby2 | hrm, there has to be a reason why the wrong context is getting used:( |
| 01:19.59 | dssman | always a reason :) |
| 01:20.11 | *** join/#asterisk BeeBuu (n=beebuu@59.38.99.2) |
| 01:20.59 | scooby2 | The weird thing is is that it is random. Sometimes it transfers to the right context. Others is does not. |
| 01:22.17 | *** join/#asterisk sasargen (n=chatzill@68-245-101-122.pools.spcsdns.net) |
| 01:30.42 | *** join/#asterisk moy (n=moy@201.54.232.194) |
| 01:36.59 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 01:38.46 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 01:46.12 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:48.22 | *** join/#asterisk japerry (n=japerry@drupal.org/user/45640/view) |
| 01:53.11 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 01:53.56 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 02:01.26 | *** join/#asterisk syntxerr (n=no@CPE001d7e602900-CM0011aec52a9c.cpe.net.cable.rogers.com) |
| 02:04.10 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 02:04.37 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 02:05.57 | *** join/#asterisk sprite-- (n=sprite@12.228.1.131) |
| 02:08.59 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.3) |
| 02:10.32 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 02:15.36 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 02:16.29 | *** join/#asterisk andresmujica (n=andresmu@190.27.101.204) |
| 02:17.56 | *** join/#asterisk Skavin (i=kevins@202-180-69-27.me.tranzpeer.net) |
| 02:25.05 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 02:32.16 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-e895b2a533fe0be2) |
| 02:38.48 | *** join/#asterisk andresmujica (n=andresmu@190.27.101.204) |
| 02:39.12 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 02:42.03 | Skavin | Anyone having issues with asterisk when clients are using a=ptime:30 |
| 02:46.53 | Skavin | I have asterisk playing out the rtp packets at steps of 20 40 20 40 and receving 20 ms on leg a and 30 0 30 30 0 30 and receving 20 ms on leg b |
| 02:47.29 | Skavin | sorry leg A is reveving at 30ms intervals |
| 02:47.47 | Skavin | How do I get asterisk to play out with no jutter |
| 02:48.03 | Skavin | its not receving teh packets with jitter |
| 03:00.01 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:09.27 | grndslm | you guys think a grandcentral invite is worth $125? |
| 03:09.51 | seanbright | note to self: logical span numbers are, in fact, relevant when configuring NFAS |
| 03:10.06 | jql | that's true |
| 03:10.28 | seanbright | this gem brought to you by: the guy who just spent 2 hours on the phone with telco |
| 03:10.31 | seanbright | heh |
| 03:13.42 | grndslm | AHEM... anybody think a grandcentral invite is worth $125? |
| 03:13.51 | *** join/#asterisk ShaneAu (n=shane@203.56.250.55) |
| 03:13.54 | ShaneAu | hey |
| 03:13.57 | jql | can't ask me... I work for a phone company |
| 03:14.44 | ShaneAu | I'm having trouble understanding why asterisk logs RINGNOANSWER in queue_log when all members part of that particular queue are already on a call./ |
| 03:14.49 | ShaneAu | Eg. Say ther' |
| 03:14.58 | [TK]D-Fender | seanbright: this is telecom.... logic does not come bundled with the package ;) |
| 03:15.35 | ShaneAu | Say there's one member part of a queue, he/she receives a phone call, answers it, another call comes in while they're still on the phone, I see a heap of "RINGNOANSWER" entries. |
| 03:15.35 | seanbright | i'll learn one of these days :) |
| 03:15.57 | ShaneAu | For that call, with 0 seconds. |
| 03:16.04 | grndslm | [TK]D-Fender & drmessano: you guys think a grandcentral invite is worth $125? |
| 03:16.08 | ShaneAu | 1228360331|1228360320.82046|31|Local/5050@from-internal/n|RINGNOANSWER|0 |
| 03:16.41 | drmessano | grndslm: $125 for what would equate to 20 months of incoming phone service from a real provider? No |
| 03:16.56 | ShaneAu | How can I stop it from logging RINGNOANSWER when everyone's on the phone, I have skip busy agents on - the phone doesn't ring on a second line or anything like that. |
| 03:17.00 | grndslm | drmessano: what provider is this? |
| 03:17.11 | drmessano | grndslm: Most of them |
| 03:18.25 | drmessano | Thats like 3 years of Gizmo5 incoming |
| 03:18.30 | drmessano | No |
| 03:18.33 | drmessano | 3 1/2 |
| 03:20.05 | seanbright | i need to borrow someone who can make > 23 simultaneous calls to a toll free |
| 03:20.34 | jql | only for a prank, man. only for a prank |
| 03:20.35 | jql | :) |
| 03:20.44 | seanbright | to test, actually. |
| 03:21.31 | *** join/#asterisk heison (n=heison@i209-195-69-115.cia.com) |
| 03:21.56 | [TK]D-Fender | ~itsplist-us |
| 03:21.57 | 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 |
| 03:22.06 | grndslm | drmessano: Gizmo doesn't do the same things GC does, tho, right? |
| 03:22.35 | drmessano | grndslm: I am talking about a $35 "callin" or whatever they call it USA DID |
| 03:22.43 | drmessano | One year, $35 |
| 03:23.02 | grndslm | drmessano: but GC + gizmo is more powerful than gizmo alone, right? |
| 03:23.08 | drmessano | ... |
| 03:23.15 | drmessano | powerful? |
| 03:23.19 | grndslm | i'm liking GC... i did pay $125 for it, like a sucker |
| 03:23.21 | grndslm | i'm impatient |
| 03:23.31 | drmessano | Good god |
| 03:23.41 | drmessano | Thats like 2 years of service from an ITSP |
| 03:24.05 | grndslm | well, as long as google doesn't take it down... i'll be fine |
| 03:24.11 | heison | does anyone here use SCCP with Asterisk? |
| 03:25.52 | grndslm | drmessano: i prolly wouldn't have gone with any other ITSP besides my local cable provider or voicepulse... but I ⥠google |
| 03:26.55 | drmessano | Um ok.. |
| 03:27.07 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 03:28.59 | grndslm | drmessano: i was also looking thru this channel and saw where you said there wasn't an all-in-one router & ata... but there is one linksys device that openwrt runs on and the ata does work |
| 03:29.05 | grndslm | just don't remember the model no., tho |
| 03:30.26 | grndslm | one router with openwrt & asterisk could handle a basic residential line no problem, right? |
| 03:31.15 | drmessano | OpenWRT supports the router model, not the FXS ports |
| 03:32.18 | grndslm | =-O |
| 03:33.30 | grndslm | one dude in the OWRT channel a few weeks back said his FXS port was indeed working |
| 03:34.47 | drmessano | "OpenWRT may be capable of running, but it's still very preliminary for this platform. You also won't have any of the phone features at all in OpenWRT. So far, the FXS ports are not yet working on anything but the Linksys firmware." |
| 03:34.57 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 03:35.58 | drmessano | Did you get the info from him how he got it working? |
| 03:37.28 | grndslm | sure didn't... he assured me it worked |
| 03:37.32 | drmessano | Why not skip all this mickey mouse stuff and get vonage |
| 03:37.38 | grndslm | i wasn't planning on getting another router at the time |
| 03:37.48 | grndslm | don't wanna pay $25/mo... that's why |
| 03:38.01 | grndslm | plus taxes & fees |
| 03:38.03 | grndslm | that's BS |
| 03:38.51 | grndslm | drmessano: i think the OWRT wiki is a little dated... perhaps the guy was using svn? |
| 03:38.59 | *** join/#asterisk etfonhomey_ (n=chatzill@mobile-166-214-222-236.mycingular.net) |
| 03:39.37 | drmessano | Well, you're talking about putting asterisk on a platform many would consider non-optimal and more a proof of concept, you're in love with Google so you're looking to throw $125 into a grandcentral number piped to a Gizmo account to your box (which creates like 3 points of failure), and you still haven't addressed outbound calls, unless you're gonna do all your calling from the GC website button |
| 03:40.23 | drmessano | Sounds like you're better off with Vonage or Comcast triple play.. it will actually work reliably in the end |
| 03:41.01 | grndslm | drmessano: thing is... residential line isn't that important for my mother and i |
| 03:41.45 | grndslm | we have 2 sprint cell phones with 500 anytime minutes, n&w starting at 7pm, m2m, unlimited web, texts, etc.... for $55/mo AFTER taxes & fees |
| 03:41.51 | *** join/#asterisk ZaVoid (n=zavoid@nj-76-6-39-193.dhcp.embarqhsd.net) |
| 03:42.36 | grndslm | i'm thinking of adding the GC + gizmo line simply because i'm always close to my 500 anytime minutes every month, and i'm tired of watching my minutes |
| 03:42.49 | grndslm | i guess i'm not really thinking anymore... it's beyond that |
| 03:43.06 | grndslm | i was gonna pay $230 for an ooma box, and *they* won't be in biz longer than a year |
| 03:43.55 | drmessano | What if Google cuts off Gizmo5 support tomorrow? |
| 03:45.03 | drmessano | or what if Gizmo blocks GC? |
| 03:45.16 | grndslm | drmessano: i'd still have UNLIMITED calling from my cell phone by adding a sprint to home service for $4/mo |
| 03:45.26 | grndslm | still no need to worry about minutes |
| 03:45.32 | grndslm | GC rox my sox off |
| 03:45.41 | drmessano | So why even worry about Asterisk? or Grandcentral? |
| 03:45.48 | grndslm | for fun, silly |
| 03:46.01 | grndslm | what else will i do with my UNLIMITED time |
| 03:48.07 | *** join/#asterisk sasargen (n=chatzill@68.244.136.19) |
| 03:48.31 | grndslm | why setup mythtv & pay $20/yr for channel listings when i could just get a tivo and pay $15/mo!?!?! |
| 03:49.42 | jblack | To get hdtv. |
| 03:49.50 | jblack | For non-broadcast channels. |
| 03:50.47 | grndslm | most hd channels over cable are open for tuners like the hdhomerun |
| 03:50.52 | grndslm | still no need for a tivo |
| 03:50.59 | drmessano | most? |
| 03:51.02 | drmessano | No, a couple |
| 03:51.05 | grndslm | if not all |
| 03:51.19 | grndslm | same hd channels in my neck of the woods are available ota and thru cable |
| 03:51.21 | drmessano | Comcast has maybe a dozen |
| 03:51.25 | sprite-- | Any recommendations for termination? Would like pay per call unlimited channels with local and 800 DID |
| 03:51.31 | grndslm | cableone has half a dozen if that |
| 03:51.33 | drmessano | HD broadcast |
| 03:51.35 | [TK]D-Fender | ~itsplist-us |
| 03:51.35 | 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 |
| 03:51.37 | [TK]D-Fender | ^^^^^^^^^ |
| 03:52.03 | [TK]D-Fender | sprite--: So far I'd knock NuFone & Broadvoice off tthat list for your needs... |
| 03:52.07 | jblack | The vast majority of hdtv channels are encrypted qam. Typical exceptions are broadcast, cspan, and religious channels. |
| 03:52.13 | jblack | You asked why, that's why. |
| 03:52.23 | drmessano | Im not sure how these even compare |
| 03:53.22 | drmessano | Youre comparing using a HD tuner card to grab OTA channels + $20/yr channel listing versus a TIVO which is absolutely stupid to use for your half dozen OTA channels |
| 03:53.28 | grndslm | jblack: i'm just tellin' ya what i've heard from the locals. i have no interest in hd recording anyway... takes up too much space. i don't have 4 gigs of free space to waste on something i'm only going to delete a second after watching |
| 03:53.37 | drmessano | Why would you get a TIVO for OTA? |
| 03:53.49 | grndslm | can you even use a tivo for ota? |
| 03:53.56 | drmessano | You mentioned it |
| 03:54.02 | grndslm | i was talking about cable |
| 03:54.18 | grndslm | pretty sure the cable companies here don't use qam |
| 03:54.34 | drmessano | Over cable you get a dozen or less HD channel unencrypted |
| 03:54.40 | jblack | You're completely behind the current state of affairs, grndslm. |
| 03:54.41 | drmessano | Most are local broadcast |
| 03:54.49 | grndslm | jblack: do you have cableone? |
| 03:55.05 | jblack | I have a tivo, _and_ mythtv. |
| 03:55.18 | grndslm | i'm tellin' ya... there's maybe 6 channels channels here that are hd |
| 03:55.24 | drmessano | .... |
| 03:56.00 | jblack | this is getting nowhere. |
| 03:56.03 | drmessano | Analog on Cable is going away.. and they wont be sending unencrypted HD or standard digital channels |
| 03:56.30 | jblack | with the exception of broadcast, cspan and religious channels. ;) |
| 03:56.34 | drmessano | yeah |
| 03:56.41 | grndslm | drmessano: when is analog cable going away? |
| 03:56.41 | drmessano | <PROTECTED> |
| 03:56.54 | drmessano | As fast as the cable companies can get rid of it.. |
| 03:57.12 | drmessano | Without customer disruption |
| 03:57.15 | grndslm | took'em 4 yrs longer than they expected to get rid of analog broadcast alone!! |
| 03:57.23 | drmessano | Thats _broadcast_ |
| 03:57.28 | drmessano | Cable is a different animal |
| 03:57.35 | jblack | my local cablecompany no longer accepts analog customers. |
| 03:57.41 | drmessano | Same here |
| 03:57.43 | grndslm | yea, everybody uses cable and doesn't want to buy a new expensive hd set |
| 03:57.48 | drmessano | No |
| 03:57.54 | drmessano | Comcast gives out free boxes, as do most |
| 03:58.00 | grndslm | true |
| 03:58.01 | jblack | you can bet the big dogs like comcast, are just dying to kill analog. |
| 03:58.45 | grndslm | i don't want to pay any more money for tv whatsoever... the second analog cable's gone... we will cut it off |
| 03:58.50 | drmessano | Well, talk to a head end tech.. Analog modulators ---> 10 racks worth.. Digital modulators ----> Hmm, there's space in the 2nd rack |
| 03:59.10 | grndslm | cableone sucks... that's my problem |
| 03:59.22 | drmessano | Also no screwing with audio levels, etc |
| 03:59.31 | drmessano | Digital is a dream for operators |
| 03:59.44 | drmessano | and the boxes are a gateway to value added services |
| 03:59.50 | drmessano | On Demand, etc.. |
| 03:59.56 | drmessano | So yeah.. |
| 04:00.27 | grndslm | i see your point with comcast, but not everybody has comcast |
| 04:00.34 | drmessano | .... |
| 04:00.40 | jql | and thank goodness for that |
| 04:00.59 | drmessano | I see your point about ignoring the point and applying it to only comcast |
| 04:01.06 | jql | spits in CMCSA's general direction |
| 04:01.08 | drmessano | Im not just talking about Comcast |
| 04:01.14 | grndslm | you're talking about providers |
| 04:01.25 | grndslm | but... mississippi is slow |
| 04:01.29 | grndslm | we're the last to budge |
| 04:01.36 | drmessano | So is Georgia |
| 04:01.37 | grndslm | soo... analog cable will be here for another few moreyears |
| 04:01.42 | drmessano | and no it wont |
| 04:01.46 | grndslm | noo... they're the 10th to last to budge |
| 04:01.57 | drmessano | Are you ever wrong about anything? |
| 04:02.05 | grndslm | are YOU? |
| 04:02.17 | grndslm | that's how your tiff with the guy asking about an all-in-one router & ata started |
| 04:02.19 | drmessano | Yeah.. often.. |
| 04:02.31 | grndslm | that dude was a pud, tho... |
| 04:02.38 | jblack | grndslm: You're wrong. You're not even close. |
| 04:02.39 | drmessano | Are you some kind of stalker? |
| 04:02.53 | grndslm | some kind |
| 04:03.14 | grndslm | do you have any sources on the end of analog cable? |
| 04:03.18 | jblack | wonders what will happen when the general public figures it out. |
| 04:03.53 | grndslm | there's other linux cards that ignore the qam flag, tho... so either way... mythtv 4 life, dawgz! |
| 04:04.06 | drmessano | grndslm: Yeah, people in the industry |
| 04:04.18 | grndslm | a link, maybe? |
| 04:04.23 | drmessano | To friends? |
| 04:04.29 | drmessano | How does that work? |
| 04:04.30 | jblack | grndslm: The _stream_ is encrypted. No cablecard, no decryption. There are exactly 0 tv cards for linux (and about 1 for windows) that support cablecard. |
| 04:04.37 | [TK]D-Fender | heading out. Later all |
| 04:04.40 | jblack | grndslm: And that's how it will remain. |
| 04:04.57 | drmessano | jblack: You can ignore the encrpytion, dawg |
| 04:05.13 | jblack | sure, if you're willing to ignore what's in it -- the shows. :) |
| 04:05.14 | drmessano | jblack: ignore=very_dawg |
| 04:05.39 | jblack | but your sarcasm is noted. |
| 04:05.44 | drmessano | lol |
| 04:05.47 | grndslm | we'll see |
| 04:06.10 | grndslm | back on topic... what softphones are popular for linux? |
| 04:06.11 | jblack | ok. you do that. |
| 04:07.56 | drmessano | goes back to trying to stick it to _the man_ and run his fridge off the mad free telephone line power hack |
| 04:08.00 | jblack | drmessano: You wouldn't happen to know offhand what the effective bit depth for cablecard's encryption is, would you? |
| 04:08.39 | grndslm | Analog Cable Subscribers â The latest FCC rule, adopted on September 11, 2007, will allow continued access to local stations for cable customers (including those with standard analog cable service) following the transition to digital TV in 2009. This rule applies only to local television stations, not cable networks or premium cable channels and the rule expires in 2012. That being said, almost every MSO (including Advance/Newhouse pr |
| 04:08.39 | grndslm | operties) plans to continue analog cable service after the transition. How long will it last? The simple answer is, âas long as it makes good business sense.â Which may be a very long time. |
| 04:08.45 | jblack | Yeah. Hook a generator up to your bathroom sink, and turn on the tap for electricity. |
| 04:08.51 | Maliuta | drmessano: you're running the power from an fxo line through a converter to give you a viable 240v system? |
| 04:09.41 | jblack | limitless free energy! if only the water bill hadn't gone up to three hundred bucks, that 50 bucks of free power would be great! |
| 04:09.48 | drmessano | jblack: Not sure on the encryption details |
| 04:10.13 | grndslm | hope it's 4 or less |
| 04:10.24 | drmessano | grndslm |
| 04:10.26 | jblack | here's to hoping DVD Jon gets to missing HBO. |
| 04:10.27 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 04:10.29 | drmessano | To quote the above |
| 04:10.32 | drmessano | broadcast, cspan and religious channels |
| 04:11.02 | drmessano | So you'll have access to broadcast channels over cable unencrypted..... and? |
| 04:11.11 | grndslm | yea... i actually didn't read the middle of what i just copied |
| 04:11.18 | jblack | and all the barney frank you can handle. |
| 04:11.20 | grndslm | i like to skim... apparently too much |
| 04:11.30 | grndslm | Eff cspan |
| 04:12.00 | drmessano | Seems like you like to listen to/read only what you want to hear/see |
| 04:12.02 | grndslm | alright, i'm switching to tivoHD.. you guys win |
| 04:12.04 | drmessano | But anyway |
| 04:12.17 | jblack | I heard some of the cable cos have been playing with the analog output switch. |
| 04:12.40 | jblack | grndslm: Who cares what you do, other than you? I was simply answering your "why" question. |
| 04:14.16 | drmessano | Actually, if you're on Comcast.. where they claim to charge $4 a month for an extra digital box, tell them you're gonna switch to satellite because the boxes are free.. and they'll give you years free service for as many boxes as you need |
| 04:14.32 | drmessano | Its not like its costing them anything |
| 04:14.39 | drmessano | Its a win-win |
| 04:15.06 | grndslm | drmessano: thanks for the tip |
| 04:16.13 | grndslm | well...i'm about to go spend this $40 i just found... |
| 04:16.24 | grndslm | i'll stalk ya later, drmessano! MUHAHAHAH |
| 04:16.38 | drmessano | um ok |
| 04:17.38 | jblack | Comcast and such probably get a discount on their feeds for playing ball. |
| 04:17.50 | ricko73 | zomg Comcastic! |
| 04:18.21 | ricko73 | I wonder if I can play the same game with Charter |
| 04:18.27 | drmessano | Well |
| 04:18.31 | drmessano | Think about it |
| 04:19.00 | ricko73 | They've continued raising their "extended basic" price and reducing the number of channels (or rather moving the interesting ones to Digital only) |
| 04:19.38 | drmessano | If Comcast reduces service calls for noisy analog pictures and say a few amplifiers losing picture in one end of the band or another, particularly high band channels, where the digital is all low band and cheaper to amplify, they save cost, manpower, etc |
| 04:20.06 | jblack | I came up with a brilliant strategy around all their plans for break-fair-use-this, unreasonable-fee-that rent-equipment-theother.... |
| 04:20.07 | drmessano | So losing "$50 a year" for a box rental versus even one trip from a tech balances out |
| 04:20.24 | jblack | And it's brilliant, because it even gets me past the forced-to-watch-commercials crap too. |
| 04:20.34 | jblack | You ready for it? |
| 04:20.38 | jblack | I watch less TV. |
| 04:20.38 | drmessano | yeah |
| 04:21.14 | ricko73 | wonders how much a pain it will be to get any of the digital boxes to work with MythTV |
| 04:21.19 | jblack | They're only getting my money if they sell me something I'm willing to pay for. It's like... I'm in charge. |
| 04:21.33 | ricko73 | wife really likes the way Mythtv works |
| 04:21.57 | drmessano | jblack: Maybe grndslm has a hack for getting mad WRT54G TV over an old push button convertoer box using Twitter and Netscape 4.7 |
| 04:22.10 | drmessano | PM him if you wish, although he'll read this log later |
| 04:22.24 | grndslm | hehe.. i'm reading it right now |
| 04:22.34 | grndslm | knockin' down this horrible whiskey first |
| 04:22.49 | jblack | ricko73: Depends on where they go with HDCP. |
| 04:23.37 | drmessano | ricko73: thats where cablecard comes in.. the set top digital box is gonna spit out analog |
| 04:23.44 | jblack | There's been a few "accidents" where analog outputs were disabled for shows. The more paranoid think they were in-field testing. |
| 04:23.46 | drmessano | or digital if you have a fancy DVR |
| 04:25.24 | ricko73 | hmmm. I'll have to see if any of my friends have their digital boxes |
| 04:25.41 | ricko73 | I could also get that ATT Uverse thing, but I absolutely dispise ATT |
| 04:26.03 | *** join/#asterisk joako (n=joako@adsl-074-170-252-213.sip.gnv.bellsouth.net) |
| 04:26.14 | ricko73 | despise too |
| 04:31.36 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 04:33.27 | drmessano | I just want a Shiny PBX with a big apple on it |
| 04:33.33 | drmessano | Just.... becuz |
| 04:33.57 | ricko73 | Red Delicious or Granny Green? |
| 04:34.07 | drmessano | Hmmmm |
| 04:34.24 | drmessano | If I go with green, I might be accused of running trixbox |
| 04:34.24 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 04:34.36 | *** join/#asterisk a1fa (n=fiddy@unaffiliated/a1fa) |
| 04:34.52 | a1fa | what is the transfer function for unattended transfers? |
| 04:35.25 | ricko73 | blind transfers? |
| 04:35.28 | a1fa | yes |
| 04:36.47 | ricko73 | so what were you asking? |
| 04:37.07 | a1fa | phone call comes in, i want to transfer it to a different number |
| 04:37.23 | ricko73 | why not use the transfer feature on the phone? |
| 04:37.35 | a1fa | because i want it to be un-attended |
| 04:37.41 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 04:37.50 | ricko73 | if you want to use the transfer 'feature' in asterisk, you'll need to look at 'features.conf' |
| 04:38.02 | ricko73 | core show features |
| 04:38.35 | a1fa | blindxfer => #1 |
| 04:38.57 | a1fa | so yo would dial #1p+number? |
| 04:39.01 | ricko73 | ok so press #1 (very quickly) then the extension you wish to transfer the call to |
| 04:39.38 | a1fa | i want this to happen in dialplan |
| 04:39.45 | ricko73 | ? |
| 04:39.55 | ricko73 | you don't need to 'transfer' anything then |
| 04:40.11 | ricko73 | Just use a 'goto' statement |
| 04:40.12 | a1fa | just use Dial? |
| 04:40.24 | ricko73 | where are you trying to send the call? |
| 04:40.53 | a1fa | to a different sip carrier |
| 04:41.12 | ricko73 | yeah, then you'd need a dial statement |
| 04:41.16 | [TK]D-Fender | a1fa: You're running SPA's right? |
| 04:41.43 | a1fa | yeah |
| 04:41.51 | ricko73 | just so we understand, you want an incoming call to go out directly to another sip provider without any user intervention? |
| 04:42.04 | [TK]D-Fender | a1fa: You say "when a call comes in". So far that doesn't imply you even want to RING a phone on the inside. |
| 04:42.17 | a1fa | ricko73: it doesnt sound so bad |
| 04:42.28 | a1fa | yeah.. i want it to go to my cellphone |
| 04:42.35 | a1fa | i dont want my SPA to ring |
| 04:42.37 | [TK]D-Fender | a1Are you looking to ring a phone, answer and then DECIDE to blind transfer it out to the PSTN? |
| 04:42.45 | a1fa | so i want to transfer a call to my cellphone |
| 04:42.51 | [TK]D-Fender | a1fa: then FS this has NOTHING to do with "transfering" a call at all |
| 04:43.11 | a1fa | i dont know why i thought it did :( |
| 04:43.14 | [TK]D-Fender | a1fa: call comes in -> Dial out to your other provider. End of story |
| 04:43.20 | a1fa | i am doing that now |
| 04:43.21 | ricko73 | you want to answer() a call, then dial() another number or better yet, look at followme() |
| 04:43.27 | a1fa | but it also carries my CALLER-ID |
| 04:43.32 | a1fa | so i cant see who is calling me |
| 04:43.39 | [TK]D-Fender | ricko73: that is a retarded app that never should have been made |
| 04:43.47 | [TK]D-Fender | ricko73: no better than basic dialplan. |
| 04:44.17 | [TK]D-Fender | a1fa: Does your provider SAY they support you rigging the CID? |
| 04:44.30 | a1fa | i am not sure |
| 04:44.33 | a1fa | its Broadvoice |
| 04:44.36 | [TK]D-Fender | a1fa: Get sure. |
| 04:44.38 | a1fa | hides ;) |
| 04:45.13 | a1fa | my caller-id is blocked |
| 04:45.20 | a1fa | thats what i get on my cellphone |
| 04:46.23 | a1fa | "Calling Line ID Delivery Blocking" |
| 04:46.26 | jaytee | this is soooo cool!!! http://greenbuildingelements.com/2008/12/01/hand-build-an-earth-sheltered-house-for-5000/ |
| 04:46.40 | jaytee | Hobbits! |
| 04:47.08 | a1fa | with "Calling Line ID Delivery Blocking" disabled, it just uses my caller-ID on DIAL() |
| 04:50.52 | [TK]D-Fender | a1fa: in your outbound peer do "sendrpid=yes" and comment out "fromuser=" if present. On NORMAL dials set the CID for your line as normal |
| 04:52.08 | [TK]D-Fender | jaytee: doesn't look too strdy, secure, or locatable in the case of oh... I dunno... SNOW |
| 04:52.28 | [TK]D-Fender | jaytee: falls into the "geologically sensitive" range |
| 04:52.34 | jaytee | not much snow in Wales I bet |
| 04:52.40 | a1fa | handle_response_invite: Received response: "Forbidden" from |
| 04:52.41 | jaytee | more like lots of rain |
| 04:53.02 | [TK]D-Fender | jaytee: Your home will BECOME a mud-slide! |
| 04:53.04 | jaytee | and I could go barefoot and never have to shave my feet again :-) |
| 04:53.28 | [TK]D-Fender | jaytee: the fact you do now scares me in its own right |
| 04:53.35 | jaytee | a cozy place to have Second Breakfast or Elevensies |
| 04:54.29 | [TK]D-Fender | "This just in : the picketers have broken rank and have attacked to Tetley Tea-folk in their stronghold. More to follow...." |
| 04:55.04 | jaytee | hahaha |
| 04:56.46 | [TK]D-Fender | oh noes! |
| 04:56.46 | drmessano | HA |
| 04:56.51 | [TK]D-Fender | scatters |
| 04:56.52 | drmessano | I was reading that article |
| 04:56.56 | drmessano | and then I was all like.. |
| 04:57.01 | drmessano | "Oh, british" |
| 04:57.13 | a1fa | [TK]D-Fender: i am getting Forbidden message |
| 04:57.21 | drmessano | Mr Bean car with 3 wheels <-- nuff said |
| 04:58.08 | [TK]D-Fender | a1fa: try putting the fromuser back if previously present |
| 05:00.11 | a1fa | did |
| 05:00.29 | a1fa | its using fromuser as caller id |
| 05:01.24 | a1fa | Set(CALLERID(all)=<number>) |
| 05:02.42 | *** join/#asterisk sdaniels (n=chatzill@cpe-72-190-76-209.tx.res.rr.com) |
| 05:03.22 | sdaniels | what function should I use if i want to listen for a caller to enter some numbers, and set those numers to a variable to use later? |
| 05:03.29 | [TK]D-Fender | a1fa: So you tried with/without fromuser enabled? |
| 05:03.55 | [TK]D-Fender | sdaniels: "core show application read" |
| 05:04.00 | sdaniels | thx |
| 05:04.38 | a1fa | yeah |
| 05:04.40 | a1fa | same thing |
| 05:04.44 | a1fa | forbidden |
| 05:05.38 | [TK]D-Fender | a1fa: Could be they simply don't let you.... |
| 05:05.50 | [TK]D-Fender | a1fa: BV is the "White-bread" of the VoIP world.... |
| 05:06.31 | jaytee | sdaniels, appendix B of the book is chock full of goodies like Read() and stuff like that. |
| 05:07.05 | [TK]D-Fender | jaytee: Easily misconstrued Digg article : "The miracle teabag: Stem cells in a pack help stroke victim " |
| 05:07.30 | jaytee | I saw that earlier |
| 05:07.42 | jaytee | hehehe |
| 05:08.25 | sdaniels | lol, i looked away at the TV then came back to do a double take when all i saw was "the miracle teabag" |
| 05:09.13 | drmessano | HA |
| 05:09.25 | drmessano | Isn't that when.... |
| 05:10.55 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-209-183.phlapa.east.verizon.net) |
| 05:11.00 | jaytee | I used to think I was keeping up with the times but it was just 2 years ago that I found out what a "dirty sanchez" was. |
| 05:11.36 | drmessano | Google for "Hot carl" |
| 05:11.39 | drmessano | I'll wait |
| 05:12.11 | jaytee | omg! |
| 05:12.13 | *** join/#asterisk harry_v (n=pcsuppor@S010600a0c93f6f7e.vs.shawcable.net) |
| 05:12.17 | drmessano | Life can be a cruel mistress |
| 05:12.35 | jaytee | or a sick psychotic bitch |
| 05:13.28 | drmessano | You're just one night of depressed public drinking away from creating a new slang word |
| 05:15.26 | jaytee | ah! it's also known as a "Cleveland Steamer" |
| 05:15.53 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 05:16.13 | drmessano^ | hmmm |
| 05:16.18 | drmessano^ | SOB |
| 05:16.21 | jaytee | ah, you missed it |
| 05:16.25 | drmessano^ | What? |
| 05:16.38 | jaytee | I just found out it's also known as a "Cleveland Steamer" |
| 05:16.43 | drmessano^ | LOL yeah |
| 05:17.56 | jaytee | jbot seen jaytee |
| 05:17.58 | jbot | jaytee is currently on #asterisk (40m 18s). Has said a total of 17 messages. Is idling for 2s, last said: 'jbot seen jaytee'. |
| 05:18.37 | jaytee | sabotaged my own setup |
| 05:19.24 | jaytee | I was just thinking if a prospective employer ever grepped the logs of this IRC channel and knew my IRC nick it would definitely not improve my chances. |
| 05:20.16 | a1fa | whats is the variable of person calling CID? |
| 05:20.35 | jaytee | what? |
| 05:21.23 | jaytee | a1fa, if I understand you correctly (and I'm just guessing here) you want to know the variable used to hold caller id info. |
| 05:21.23 | [TK]D-Fender | a1fa: "core show function CALLERID" |
| 05:21.34 | *** join/#asterisk brut- (n=brut@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 05:22.06 | a1fa | [TK]D-Fender: i mean, how do i get CID from user who called? |
| 05:22.16 | [TK]D-Fender | a1fa: "core show function CALLERID" |
| 05:23.00 | a1fa | get ;) |
| 05:23.03 | a1fa | dam |
| 05:23.23 | jaytee | get the variable? and do what with it? |
| 05:24.18 | a1fa | nothin' |
| 05:24.50 | a1fa | nvm |
| 05:24.54 | a1fa | it should already be set |
| 05:25.08 | a1fa | i was writing an email to broadvoice support |
| 05:25.09 | a1fa | oh well |
| 05:25.14 | a1fa | it may be time to look for another provider |
| 05:25.44 | a1fa | i've used broadvoice since 05 |
| 05:26.46 | a1fa | if anyone knows a provider, please speak now :P |
| 05:27.14 | jaytee | ~itsplist-us |
| 05:27.15 | 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 |
| 05:27.56 | a1fa | cool |
| 05:28.07 | drmessano^ | I think my 6 WRT54Gs are ARPing my network out |
| 05:29.11 | jaytee | well that sucks |
| 05:29.17 | jaytee | what can you do about that? |
| 05:29.23 | a1fa | arping? |
| 05:29.30 | a1fa | are you sure its the gateway |
| 05:29.49 | a1fa | teliax charges for inbound minutes too? |
| 05:30.00 | sprite-- | What's the proper way to make outgoing calls through Teliax? I tried exten => _X.,1,Dial(SIP/teliax/${EXTEN}) |
| 05:30.30 | sprite-- | and Dial(SIP/${EXTEN}/teliax) |
| 05:30.58 | drmessano^ | I can not connect them all at the same time in a big loop |
| 05:31.03 | drmessano^ | Hmmm |
| 05:31.06 | *** join/#asterisk bobnormal (n=irc@221.213.47.10) |
| 05:32.05 | drmessano^ | yay |
| 05:33.04 | drmessano^ | Yeah, drmessano, suck it |
| 05:33.13 | drmessano | steals his nick |
| 05:38.51 | drmessano | hmm |
| 05:41.55 | hi365 | drmessano: whats the link to the justfuckinggoogleit site? |
| 05:43.04 | ricko73 | hi365: www.phonetrace.org |
| 05:43.31 | drmessano | Looks like its down |
| 05:43.40 | hi365 | hmm |
| 05:43.44 | denon | hi365: http://letmegooglethatforyou.com/ |
| 05:43.47 | denon | rocks |
| 05:44.39 | drmessano | http://letmegooglethatforyou.com/?q=google |
| 05:44.41 | drmessano | irony |
| 05:45.13 | denon | I love that site |
| 05:45.26 | sdaniels | I have a strange issue, when a call comes in from outside, i have * answer the call, play a message, then redirect the call to an outside number, it works fine. if i do not play the message, then the caller hears the phone ringing until it is redirected, then the caller will not hear a ringing tone, but the call is successful and can be answered... any thoughts? http://pastebin.ca/1275609 |
| 05:45.53 | hi365 | ricko73: sorry, that site is blocked by the filters here |
| 05:46.02 | ricko73 | as well it should be |
| 05:46.12 | ricko73 | it's similar to meatspin.org |
| 05:46.24 | ricko73 | you don't want to go to either |
| 05:47.42 | sdaniels | tubgirl.com |
| 05:48.01 | ricko73 | you kids play, I'm going to bed |
| 05:50.06 | [TK]D-Fender | Yup... checkout time here also... later all |
| 06:00.50 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 06:03.44 | LeddyHM | what are the most common file extensions for voicemails? .wav, .mp3? Is there a .gsm, .alaw, .ulaw? |
| 06:06.08 | sdaniels | clear |
| 06:16.34 | harry_v | same here |
| 06:34.16 | *** join/#asterisk lucky|aba (n=lucky@ip72-205-200-133.sb.sd.cox.net) |
| 06:34.58 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 06:37.56 | *** join/#asterisk edoceo (n=edoceo@98.247.254.241) [NETSPLIT VICTIM] |
| 06:43.09 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) [NETSPLIT VICTIM] |
| 06:45.25 | *** join/#asterisk japerry (n=japerry@drupal.org/user/45640/view) [NETSPLIT VICTIM] |
| 06:46.36 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 06:51.34 | *** join/#asterisk mattzerah (n=matt@ozvoip.dsl.onthenet.net) [NETSPLIT VICTIM] |
| 06:51.59 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 06:52.53 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-17738374005949da) [NETSPLIT VICTIM] |
| 06:57.00 | *** join/#asterisk sdaniels (n=chatzill@cpe-72-190-76-209.tx.res.rr.com) |
| 06:57.00 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 06:58.58 | *** join/#asterisk sack (n=sack@226.Red-88-24-152.staticIP.rima-tde.net) |
| 07:02.21 | DigitalIrony | on a J1 connection to a NTT INS Net 1500 if using a pri_cpe/net what would a working switch type be for a TE121? Does it work? has anyone used it? |
| 07:03.18 | *** join/#asterisk FuriousGeorge (n=brian@ool-4354d18c.dyn.optonline.net) |
| 07:03.23 | FuriousGeorge | hey all |
| 07:03.51 | FuriousGeorge | whats that free MoH for asterisk that sounds like it could be a guitar |
| 07:03.57 | FuriousGeorge | not the classical stuff |
| 07:04.08 | FuriousGeorge | it's not with the defaul MoH is it? |
| 07:06.44 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-c8aed9065a3596f4) |
| 07:09.33 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 07:13.51 | *** join/#asterisk saftsack (n=oliver@g227021223.adsl.alicedsl.de) |
| 07:14.25 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:15.34 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 07:27.13 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
| 07:34.09 | *** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 07:38.13 | *** join/#asterisk oej_ (n=olle@ns.webway.se) |
| 07:39.40 | *** part/#asterisk FuriousGeorge (n=brian@ool-4354d18c.dyn.optonline.net) |
| 07:45.07 | *** join/#asterisk jeffspeff (i=Administ@c-98-211-62-9.hsd1.ky.comcast.net) |
| 07:55.04 | angryuser | is it still impossible to define which trunk to use for every peer using freepbx latest ? |
| 08:08.02 | *** join/#asterisk xrmx__ (n=user@host215-175-dynamic.7-79-r.retail.telecomitalia.it) |
| 08:08.50 | ShaneAu | When running the show queues command, you get a XXs holdtime value, which appears to be an average. |
| 08:08.54 | ShaneAu | What is the average over? |
| 08:14.48 | hesco | Any experience here with an AG-188N ??? |
| 08:19.58 | *** join/#asterisk Daejeo (n=chatzill@118.221.248.136) |
| 08:23.21 | *** join/#asterisk Kumbang (n=dsp@167.205.24.69) |
| 08:26.58 | *** join/#asterisk johnf (n=johnf@124-168-145-163.dyn.iinet.net.au) |
| 08:35.41 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 08:36.55 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 08:45.33 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 08:56.46 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 08:59.33 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 09:04.08 | *** join/#asterisk chi6IT41 (n=chigital@services.mivitec.net) |
| 09:04.16 | *** join/#asterisk chi6IT41 (n=chigital@services.mivitec.net) |
| 09:21.46 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:26.00 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:34.42 | *** join/#asterisk protocols (n=protocol@p5791FD9B.dip.t-dialin.net) |
| 09:40.11 | *** join/#asterisk CrashHD (n=CrashHD@65.74.156.108) |
| 09:59.10 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 09:59.18 | dominic1 | I have a little urgent problem |
| 09:59.50 | dominic1 | my asteriskprocess is dying.... |
| 09:59.58 | dominic1 | my system has a lot of 40 |
| 10:00.11 | dominic1 | the asteriskprocess is killing anything |
| 10:00.13 | dominic1 | :-D |
| 10:00.48 | dominic1 | How can I troubleshoot asterisk to get information about which module is consuming so much cpu and ram? |
| 10:12.41 | DigitalIrony | on a J1 connection to a NTT INS Net 1500 if using a pri_cpe/net what would a working switch type be for a TE121? Does it work? has anyone used it? |
| 10:14.47 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 10:15.41 | *** part/#asterisk johnf (n=johnf@124-168-145-163.dyn.iinet.net.au) |
| 10:20.56 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 10:21.27 | *** join/#asterisk lou_gr (n=lou_gr@212-70-216-131.ath.static.tee.gr) |
| 10:24.40 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 10:32.19 | echinos | Is there a way to get asterisk to play audio to speakers connected to the server? I want to have the server announce when there are incoming calls, for example |
| 10:35.46 | dominic1 | execute a systemscript in the dialplan |
| 10:35.56 | angryuser | can someone tell me what is this ? ec 4 10:53:50 WARNING[5602]: channel.c:889 channel_find_locked: Warning: Avoided contention wait for '0xb77482c8', 10 retries! RETURN = NULL |
| 10:37.38 | echinos | dominic1: ah, ok. thx. |
| 10:37.42 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 11:06.44 | *** join/#asterisk chendy (n=chatzill@58.251.111.222) |
| 11:07.22 | *** join/#asterisk dramman (n=Miranda@122.111.59.159) |
| 11:08.31 | dramman | What's the simplest way to get voicemail working with email notifications? |
| 11:09.26 | dramman | Having a bit of trouble setting up sendmail, but I have a hosted email server, so this seems like a bit of over-kill? |
| 11:14.36 | angryuser | dramman: you have allways the solution to install not from source, something like elastix, that would require less effort ;) or why not to configure a simple server on * machine ? |
| 11:14.56 | *** join/#asterisk curious101 (i=curious1@122.54.45.14) |
| 11:17.21 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 11:18.33 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 11:18.59 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 11:19.03 | fcois93 | hello all |
| 11:19.12 | dramman | I've got the latest version of asteriskNow - I'm surprised that email, bluetooth etc aren't a lot more "out-of-the-box" |
| 11:19.26 | fcois93 | how can I do a database query in the dialplan ? |
| 11:19.28 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:20.41 | angryuser | fcois93: with sql request |
| 11:23.02 | fcois93 | angryuser: have you an example? I don't find it |
| 11:23.05 | tzafrir_laptop | dramman, you can always use a simple send-only sendmail compatible program (such as ssmtp) |
| 11:23.29 | tzafrir_laptop | Asterisk won't queue messages that are not sent, though |
| 11:23.53 | tzafrir_laptop | but why can't you use the local sendmail? |
| 11:24.31 | angryuser | fcois93: it is mysql or else ? |
| 11:25.11 | fcois93 | angryuser: mysql |
| 11:25.24 | angryuser | core show application MYSQL in cli |
| 11:25.45 | *** join/#asterisk mv2 (n=mv2@83.240.229.38) |
| 11:26.07 | mv2 | anyone already used a cisco 7961 phone with asterisk ? |
| 11:26.28 | fcois93 | angryuser: thankyou |
| 11:26.52 | *** join/#asterisk stimpie (n=stimpie@84-104-5-227.cable.quicknet.nl) |
| 11:30.28 | *** join/#asterisk proppy (n=proppy@mon75-14-88-163-176-155.fbx.proxad.net) |
| 11:30.31 | *** part/#asterisk proppy (n=proppy@mon75-14-88-163-176-155.fbx.proxad.net) |
| 11:36.09 | *** join/#asterisk axisys (i=iqbala@svalbard.freeshell.org) |
| 11:38.26 | mv2 | anyone already used a cisco 7961 phone with asterisk ? |
| 11:42.26 | *** join/#asterisk chendy (n=chatzill@121.35.145.245) |
| 11:42.50 | Maliuta | mv2: there is no point asking every 10 minutes |
| 11:43.10 | Maliuta | mv2: and have you tried google and the wiki? |
| 11:48.08 | *** join/#asterisk propellerhead (n=yogurt2u@host130.190-226-46.telecom.net.ar) |
| 11:55.17 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-11-187.w86-215.abo.wanadoo.fr) |
| 11:57.15 | miloux | mv2: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx |
| 11:57.50 | mv2 | hey miloux thanks |
| 11:57.56 | miloux | np |
| 11:58.17 | mv2 | already follow that tutorial but had a problem ... but is fixed now |
| 11:58.18 | mv2 | thanks |
| 11:58.19 | *** join/#asterisk JBenden76_ (n=JBenden7@cblmdm72-240-122-140.buckeyecom.net) |
| 11:58.19 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-11-187.w86-215.abo.wanadoo.fr) |
| 12:00.30 | *** join/#asterisk axisys (i=iqbala@otaku.freeshell.org) |
| 12:03.07 | mv2 | hey maliputa : why there is no point to ask every 10 minutes ? |
| 12:05.05 | *** join/#asterisk Muhammad (n=ZXF@unaffiliated/eng1n33r) |
| 12:10.09 | *** join/#asterisk ZaVoid (n=zavoid@nj-76-6-39-193.dhcp.embarqhsd.net) |
| 12:10.42 | *** join/#asterisk Segnale007 (n=Pietro@host179-13-dynamic.12-79-r.retail.telecomitalia.it) |
| 12:15.15 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 12:21.52 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1096590415.dsl.bell.ca) |
| 12:24.42 | *** join/#asterisk linuxstb_ (n=linuxstb@rockbox/developer/linuxstb) |
| 12:25.39 | *** join/#asterisk chendy (n=chatzill@58.61.196.58) |
| 12:35.38 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 12:38.44 | *** join/#asterisk linuxstb_ (n=linuxstb@rockbox/developer/linuxstb) |
| 12:50.59 | *** join/#asterisk ZaVoid (n=zavoid@75.147.121.177) |
| 12:55.28 | *** join/#asterisk ming_zym (n=ming_zym@59.108.136.31) |
| 12:58.19 | *** join/#asterisk axisys (i=iqbala@otaku.freeshell.org) |
| 13:01.06 | *** join/#asterisk sasargen (n=chatzill@173.100.18.91) |
| 13:03.44 | *** join/#asterisk tmjb (n=tmjb@212.200.239.230) |
| 13:04.29 | *** join/#asterisk axisys (i=iqbala@otaku.freeshell.org) |
| 13:11.10 | *** join/#asterisk axisys (i=iqbala@otaku.freeshell.org) |
| 13:23.24 | *** join/#asterisk miloux (i=milu@213.88.194.123) |
| 13:27.28 | *** join/#asterisk kerx (n=prepro@adsl-68-123-206-108.dsl.irvnca.pacbell.net) |
| 13:27.33 | kerx | hi everyone, i keep receiving: |
| 13:27.38 | kerx | "${STRFTIME(${EPOCH},,%d%mNaVH:NaVS)}" "${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}" "${UNIQUEID}" "${CDR(clid)}" "${CDR(src)}" "${CDR(dst) |
| 13:27.38 | kerx | }" "${CDR(dcontext)}" "${CDR(channel)}" "${CDR(dstchannel)}" "${CDR(lastapp)}" "${CDR(lastdata)}" "${CDR(start)}" "${CDR(answer)}" " |
| 13:27.38 | kerx | ${CDR(end)}" "${CDR(duration)}" "${CDR(billsec)}" "${CDR(disposition)}" "${CDR(amaflags)}" "${CDR(accountcode)}" "${CDR(uniqueid)}" |
| 13:27.40 | kerx | "${CDR(userfield)}") |
| 13:27.42 | kerx | exten => 1,2,Dial(SIP/grnvoip/1238040112132012400) |
| 13:27.44 | kerx | exten => 3,1,System(/root/attributed_d/cdr.pl press3-suppression "${uniqueid}" "${pnumber}" "${HANGUPCAUSE}" "${CHANNEL}" "${CONTEXT |
| 13:27.49 | kerx | }" "${STRFTIME(${EPOCH},,%d%mNaVH:NaVS)}" "${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}" "${UNIQUEID}" "${CDR(clid)}" "${CDR(src)}" "${CDR(d |
| 13:27.52 | kerx | st)}" "${CDR(dcontext)}" "${CDR(channel)}" "${CDR(dstchannel)}" "${CDR(lastapp)}" "${CDR(lastdata)}" "${CDR(start)}" "${CDR(answer)} |
| 13:27.55 | kerx | " "${CDR(end)}" "${CDR(duration)}" "${CDR(billsec)}" "${CDR(disposition)}" "${CDR(amaflags)}" "${CDR(accountcode)}" "${CDR(uniqueid) |
| 13:27.58 | kerx | }" "${CDR(userfield)}") |
| 13:28.00 | kerx | exten => 3,1,Hangup |
| 13:28.02 | kerx | shit. |
| 13:28.04 | kerx | omg.... |
| 13:28.06 | kerx | slaps himself |
| 13:28.08 | kerx | that was supposed to be a one-liner |
| 13:28.10 | kerx | let me try that again |
| 13:28.12 | kerx | hi everyone |
| 13:28.14 | kerx | i keeep receiving: |
| 13:28.18 | kerx | WARNING[27141]: chan_sip.c:12389 handle_response_invite: Received response: "Forbidden" from '"asterisk" <sip:asterisk@208.79.91.51>;tag=as75ef8629' |
| 13:28.23 | kerx | NOTICE[27150]: pbx_spool.c:355 attempt_thread: Call failed to go through, reason (8) Congestion (circuits busy) |
| 13:28.26 | kerx | Really destroying SIP dialog '3cc0162a4c27ae2f0b04052119b743ff@208.79.91.51' Method: INVITE |
| 13:41.09 | *** join/#asterisk feeds (n=feeds@85-135-244-9.adsl.slovanet.sk) |
| 13:48.06 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 13:48.30 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 13:49.47 | *** join/#asterisk jtodd (n=jtodd@nat/digium/x-73b8f3e658be92d0) |
| 13:50.45 | *** join/#asterisk Yourname`_ (i=Yourname@unaffiliated/yourname/x-837320) |
| 13:50.52 | *** join/#asterisk neurosys (n=neurosys@c-66-229-81-161.hsd1.fl.comcast.net) |
| 13:54.43 | *** join/#asterisk etfonhomey (n=chatzill@www2.askpri.org) |
| 13:55.40 | tmjb | hello i user asterisk 1.4.21 compiled from source and misdn 1.1.8 the problem is that i have one way audio from incoming calls only then i try to switch to 1.6.0.1 and evrything works great. any got idea ? |
| 13:57.51 | coppice | yeah. use 1.6.0.1 |
| 13:59.12 | jasonwoot | Can I specify 'bindaddr=0.0.0.0' outside of [general] in sip.conf? |
| 13:59.32 | drmessano | lol |
| 14:00.28 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:01.11 | *** join/#asterisk Chesther (n=cam2@cam2-win.cit.cornell.edu) |
| 14:03.35 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 14:06.34 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 14:10.03 | *** join/#asterisk pecanha (n=e@189.106.34.42) |
| 14:11.28 | *** join/#asterisk feeds (n=feeds@85-135-244-9.adsl.slovanet.sk) |
| 14:11.57 | feeds | hi, will GCC 4,3 be good for copiling asterisk? |
| 14:12.37 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:12.37 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:14.11 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:14.22 | feeds | anyone? |
| 14:15.46 | *** join/#asterisk nikko (n=nikko@69.57.49.100) |
| 14:17.34 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:19.07 | *** join/#asterisk mm_202 (n=mm_202@unaffiliated/mm-202) |
| 14:21.33 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:26.47 | Katty | hugs jaytee |
| 14:27.10 | jaytee | mornin Katty |
| 14:27.16 | jaytee | *hugs* |
| 14:29.41 | Katty | jaytee: there's a toshiba meeting here today :/ |
| 14:29.41 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 14:29.49 | Katty | jaytee: i guess it wasn't enough that they want to do asterisk and samsung |
| 14:30.05 | Katty | jaytee: now they're apparently looking into being a toshiba dealer for their phone stuff |
| 14:30.27 | Katty | jaytee: this company just sold an analog samsung system :/ |
| 14:30.31 | Katty | is doomed. |
| 14:32.40 | jaytee | you're not doomed! |
| 14:33.05 | jaytee | telnettech, hi guy! |
| 14:33.15 | telnettech | whats going on |
| 14:33.29 | jaytee | same old, same old. you? |
| 14:33.34 | telnettech | hey you know any techs that will work hard and for free? |
| 14:33.52 | telnettech | we need 10 of those to lighten the workload on me |
| 14:33.59 | jaytee | I know of a few that will do the first part but no one comes to mind for the second part :-) |
| 14:34.27 | telnettech | lol |
| 14:34.31 | jaytee | telnettech, you might try on Monster.com |
| 14:35.29 | jaytee | Katty, isn't your job more internal support? or do you do Asterisk installs and integration? |
| 14:35.34 | telnettech | no the company isnt looking to hire.....we just need a few techs that can work on the systems to clear the overwhelming workload |
| 14:36.26 | telnettech | we have 400 customers worldwide and 3 techs in office to support them....2 guys quit recently and the company says they are not hiring any "bodies" |
| 14:36.44 | jaytee | 2 of your 3 techs quit? |
| 14:36.45 | *** join/#asterisk jasonpr (n=jasonpr@c-67-161-245-201.hsd1.ut.comcast.net) |
| 14:37.12 | jasonpr | Anyone have experience recording voicemail in mp3 format? |
| 14:37.46 | jaytee | jasonpr, don't think it'll support that |
| 14:39.05 | jasonpr | this is what I get when I try |
| 14:39.06 | jasonpr | <PROTECTED> |
| 14:39.06 | jasonpr | [Dec 3 00:29:59] WARNING[2438]: file.c:1063 ast_writefile: No such format 'mp3' |
| 14:39.06 | jasonpr | <PROTECTED> |
| 14:39.07 | jasonpr | [Dec 3 00:29:59] WARNING[2438]: app.c:759 __ast_play_and_record: Error creating writestream '/var/spool/asterisk/voicemail/hotkeys/7000/tmp/WYwsPr', format 'mp3' |
| 14:39.18 | jasonpr | is there a way to write an adapter for it? |
| 14:40.16 | jaytee | jasonpr, you can use mp3 for MOH but I don't think you can use it as a format for voicemail, just wav, gsm, ulaw for sure and I'm not certain about g729 |
| 14:41.18 | jaytee | jasonpr, as for writing an adapter if you download the tarballs for * 1.4 or 1.6 you get ALL the source code so you can recode it to your hearts content :-) sounds like a fun challenge! |
| 14:43.15 | *** join/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 14:43.44 | tzafrir_laptop | jasonpr, you can generally do that with a monitor/mixmonitor script |
| 14:44.40 | tzafrir_laptop | Keep in mind that the mp3 patent holders tend to enforce it with respect to encoding mp3 |
| 14:44.57 | *** join/#asterisk almir (i=almir@91.191.20.108) |
| 14:45.41 | *** join/#asterisk mort_gib (n=mjensen@162.228.120.212.dsl.dynamic.gibconnect.com) |
| 14:46.05 | tzafrir_laptop | If that bothers you, use ogg/vorbis . IIRC asterisk supports it |
| 14:46.17 | tzafrir_laptop | or speex |
| 14:47.00 | tzafrir_laptop | generally both vorbis and mp3 are not really optimized for compressing sound. Which is why speex, gsm and such can get you smaller files |
| 14:47.36 | coppice | I think you mean not optimised for speech |
| 14:47.51 | *** join/#asterisk Karlitoo (n=proscom@213.137.110.67) |
| 14:48.00 | mort_gib | Hi I need a hand with a Polycom SoundStation ip 6000, check_auth: username mismatch, have <450> , digest has <> |
| 14:48.38 | mort_gib | I have to do this via the webiterface, yes I KNOW, but I have only that very same Polycom on the network!! |
| 14:49.13 | feeds | hi all, I'm trying to compile asterisk 1.6.0.1 on fedora 9, an I'm getting this error when I run make install : http://asterisk.pastebin.com/f123cf307 |
| 14:49.22 | coppice | IP6000 is a nice machine. it does excellent wideband voice |
| 14:49.41 | feeds | I ran make menuselect and ./configure and make clean before of course.... |
| 14:49.46 | mort_gib | I know, it really sleek |
| 14:49.55 | mort_gib | Client LOVES it :-) |
| 14:50.31 | mort_gib | Thing is I mostly use Snoms and I use the exact same setting in sip.conf but it won't register! |
| 14:50.34 | coppice | G.722.1 Annex C is really nice |
| 14:50.34 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 14:53.00 | feeds | hi all, I'm trying to compile asterisk 1.6.0.1 on fedora 9, an I'm getting this error when I run make install : http://asterisk.pastebin.com/f123cf307 | I ran make menuselect and ./configure and make clean before of course.... | Anyone? |
| 14:54.40 | hesco | Any experience here with an AG-188N ??? Or perhpas a more relevant question, I'm looking for an afforable ATA(preferably IAX2 capable)) for use behind a firewall. Any advice? I started by looking for the $45 FXS/FXO board which digium used to sell. |
| 14:58.11 | Karlitoo | I need help compiling pwlib I get a make[2]: *** [/src/pwlib_v1_9_0/lib/obj_linux_x86_r/psasl.o] Error 1 |
| 14:58.33 | Karlitoo | I've googled but with no luck on resolution |
| 14:58.40 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 14:58.45 | jasonwoot | Can I specify 'bindaddr=0.0.0.0' outside of [general] in sip.conf? |
| 14:58.49 | Karlitoo | I know it's an old pwlib but I need that specific one |
| 14:58.59 | telnettech | jaytee: sorry about that |
| 14:59.26 | telnettech | jaytee: we had 5 techs and now we only have 3 |
| 15:01.29 | *** join/#asterisk propellerhead (n=yogurt2u@host89.190-136-111.telecom.net.ar) |
| 15:02.32 | *** join/#asterisk pecanha (n=e@189.106.186.94) |
| 15:02.43 | Karlitoo | is sasl even required for pwlib to work |
| 15:02.43 | Karlitoo | ?? |
| 15:03.18 | mort_gib | I found loads of entries on google regarding the check_auth: username mismatch -Does anyone know what the fix is?? |
| 15:05.00 | jaytee | telnettech, sounds like a "race to the bottom". I'd get my resume updated and start looking in earnest or communities near earnest :-) |
| 15:05.21 | telnettech | i did that last nigt |
| 15:05.35 | telnettech | dice and monster |
| 15:05.39 | coppice | VoIP is mostly a race to the bottom. |
| 15:05.45 | telnettech | still need to do careerbuilder |
| 15:06.06 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:07.06 | Karlitoo | I need help compiling pwlib I get a make[2]: *** [/src/pwlib_v1_9_0/lib/obj_linux_x86_r/psasl.o] Error 1 |
| 15:07.11 | Karlitoo | I found http://threebit.net/mail-archive/openh323/msg00981.html |
| 15:09.21 | fcois93 | hello, I have a problem with my calls |
| 15:09.30 | fcois93 | I have : |
| 15:09.42 | fcois93 | <PROTECTED> |
| 15:09.47 | fcois93 | <PROTECTED> |
| 15:09.52 | fcois93 | <PROTECTED> |
| 15:10.07 | fcois93 | <PROTECTED> |
| 15:10.14 | fcois93 | <PROTECTED> |
| 15:10.33 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 15:10.55 | fcois93 | iasterisk1 say to asterisk 2 No user '103' in SIP users list |
| 15:10.55 | fcois93 | Found peer 'media' for '103' from 192.168.1.151:5060 |
| 15:11.05 | fcois93 | have you an idea ? |
| 15:15.09 | *** join/#asterisk Segnale007 (n=Pietro@host88-253-dynamic.42-79-r.retail.telecomitalia.it) |
| 15:15.39 | *** join/#asterisk chouchen (n=chatzill@125.115.174.133) |
| 15:19.46 | *** join/#asterisk mog (n=mog@nat/digium/x-ffcc371e13a9c840) |
| 15:19.46 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:23.00 | pecanha | hello, anyone using PAP2 with asterisk? |
| 15:25.30 | tzafrir_laptop | ~ask |
| 15:25.31 | jbot | i heard ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:29.15 | *** join/#asterisk hohum (n=dcorbe@206.71.169.115) |
| 15:30.46 | Karlitoo | what do I need to put into makefile so it doesn't load a module |
| 15:30.48 | Karlitoo | ?? |
| 15:32.16 | awk_r | Karlitoo, edit /etc/asterisk/modules.conf and follow the 'noload' samples (assuming you ma[d]e samples |
| 15:32.29 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 15:33.30 | Karlitoo | sry awk_r I wasn't specific enaugh, while compiling what do I need to put into makefile |
| 15:34.12 | awk_r | have you tried make menuselect? |
| 15:34.38 | Karlitoo | I'm compiling old asterisk |
| 15:34.41 | Karlitoo | 1.2 |
| 15:35.00 | Karlitoo | need it because h323 works on it |
| 15:35.34 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-a45525bbf5756855) |
| 15:35.34 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:37.18 | *** join/#asterisk qdk (n=qdk@212.27.24.104.bredband.3.dk) |
| 15:42.54 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 15:43.53 | *** join/#asterisk Corydon76-dig (i=peuce@pdpc/supporter/bronze/Corydon76-home) |
| 15:44.38 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 15:47.08 | etfonhomey | Does AgentCallbackLogin() still exist in 1.6? |
| 15:48.13 | jasonwoot | boy, I hope so |
| 15:48.28 | jasonwoot | or else jasonwoot is on 1.4 for perpetuity |
| 15:49.27 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:51.04 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:51.53 | *** join/#asterisk xrmx__ (n=user@host5-117-dynamic.60-82-r.retail.telecomitalia.it) |
| 15:54.28 | *** join/#asterisk Spirits-Sight (n=christop@76.118.226.101) |
| 15:55.57 | Spirits-Sight | What do people use for out going calls that include at less 4 channels and is a flat rate fee? VoicePulse changes me 0.02 cent for call near me |
| 15:56.19 | *** join/#asterisk legis (n=wadsack@unaffiliated/legis) |
| 15:56.32 | Spirits-Sight | I want to deal with a company that is respected through |
| 16:00.15 | *** join/#asterisk psy0nid3 (n=chatzill@bookit-dev.com) |
| 16:02.47 | legis | ~sipnat |
| 16:02.47 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:05.54 | *** join/#asterisk Leddy (n=Leddy@72.54.198.194) |
| 16:06.50 | telnettech | I need someone that can read SIP today |
| 16:06.55 | telnettech | http://pastebin.com/d75018d9 |
| 16:07.18 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 16:07.19 | telnettech | I dont seem to be getting MW light to work on the Grandstream GXP 2000 phones |
| 16:07.36 | telnettech | can someone look at the pastebin and tell me what I may be missing? |
| 16:08.35 | *** join/#asterisk mv2 (n=mv2@83.240.229.38) |
| 16:09.40 | etfonhomey | telnettech, does the user have 4 messages that they've already listened to? |
| 16:10.32 | mv2 | anyone already manged to put a cisco 7961 phone with nat enabled on asterisk ? |
| 16:10.41 | telnettech | no there is only 1 message in the voicemail directory for this user. which is the message i left as a test |
| 16:10.42 | mv2 | already search for it in google |
| 16:10.51 | mv2 | with no luck |
| 16:12.03 | pecanha | ? |
| 16:12.28 | pecanha | ~pap2 |
| 16:12.29 | jbot | hmm... pap2 is a Linksys ATA with 2 FXS ports typically locked to Vonage. |
| 16:12.57 | telnettech | I have the SIP Demystified book coming thru the mail to me so that I can learn more about SIP messages |
| 16:13.38 | etfonhomey | telnettech, pastebin your sip.conf entry for 505 |
| 16:14.14 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 16:14.14 | etfonhomey | telnettech, and your voicemail conf file. |
| 16:14.43 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 16:15.04 | telnettech | ET: voicemail is done a little differently....We use a MySQL table |
| 16:15.27 | telnettech | if you are looking for the context in the voicemail.conf file, I have double checked that it is correct |
| 16:15.38 | etfonhomey | :), that was #1 step. |
| 16:16.10 | etfonhomey | telnettech, pastebin the results of a sip show subscriptions |
| 16:16.22 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 16:17.45 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-2478ca248281529e) |
| 16:17.45 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:18.22 | telnettech | ET: here is the pastebin. I have included the sip.conf for this 505 as well as the mysql DB entry as well |
| 16:18.26 | telnettech | http://pastebin.com/d7cf7df0f |
| 16:18.27 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:19.37 | *** join/#asterisk clintc (n=clintc@n128-227-5-240.xlate.ufl.edu) |
| 16:19.49 | telnettech | as you can see, the mailbox parameter is equal to the database entry |
| 16:22.19 | *** join/#asterisk Yourname` (i=Yourname@unaffiliated/yourname/x-837320) |
| 16:23.48 | etfonhomey | Yep. |
| 16:23.49 | ghento | Hi all. Just curious, is there a way to test a system to see how many outgoing calls it can handle? kind of like a stress test (for sip) |
| 16:24.32 | *** join/#asterisk ickmund (n=ickmund@ada-bcn-fw01.adamoeurope.com) |
| 16:24.43 | *** join/#asterisk dahunter3 (n=dahunter@pool-72-67-222-109.lsanca.fios.verizon.net) |
| 16:24.52 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:25.56 | etfonhomey | telnettech, pastebin the sip show subscriptions results |
| 16:25.58 | dahunter3 | I've got a weird one. Phone calls from sip devices seem to work fine. However, I'm setting up click to call. It is working but when the manager does the originate it says chan_sip.c:3022 sip_call: No audio format found to offer. Cancelling call to 1200 |
| 16:26.51 | telnettech | it shows blank ET |
| 16:27.58 | etfonhomey | telnettech, the sip notify message is saying that there are 0 new and 4 old messages |
| 16:28.18 | telnettech | thats what i thought |
| 16:28.40 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 16:28.42 | etfonhomey | telnettech, I'm not sure your phone is looking for messages in the right place |
| 16:28.46 | telnettech | what does the show suscriptions supposed to give ack, voicemail users |
| 16:29.21 | etfonhomey | telnettech, if I could get to my * box, I would show you. :) |
| 16:29.52 | pecanha | hey guys, when you can call from a device, but you can't receive calls, the problem can be codec yet? |
| 16:29.54 | etfonhomey | telnettech, I believe a phone subscribes to "watch" the mailbox and you should see that information in sip show subscriptions. |
| 16:30.28 | *** join/#asterisk muhkoenig (n=farmer@port-212-202-20-173.dynamic.qsc.de) |
| 16:30.54 | etfonhomey | telnettech, I'm leaning towards the phone is not provisioned correctly. |
| 16:32.04 | telnettech | ET: cool, let me compare the phone settings with another site that I know it is working....thanks again for your help |
| 16:32.18 | fcois93 | please help me for a problem with a complex forward call with 3asterisks http://forums.digium.com/viewtopic.php?p=121425#121425 |
| 16:34.50 | fcois93 | I call through some asterisk, (asterisk1--asterisk2--asterisk3) when the call is in asterisk3, it forward it to asterisk2... but I have an error in asterisk2 which said that user '103' not found! it saw the first call passe and now, it don't know who is the use... ! |
| 16:36.56 | *** join/#asterisk Yourname`_ (i=Yourname@unaffiliated/yourname/x-837320) |
| 16:37.26 | fcois93 | nobody can help me? |
| 16:37.41 | etfonhomey | telnettech, if you can capture the SUBSCRIBE message from that phone relating to the MWI, it might shed some more light on the situation. |
| 16:37.46 | *** join/#asterisk Aces1up (n=Now@ip70-173-52-152.lv.lv.cox.net) |
| 16:37.48 | Aces1up | hey all i have a lil issue, it seems like for some reason from time to time my sangoma 200 card will get like jammed up or something.. as i cannot get any incoming calls, or make any outgoing calls on it, i then have to go into asterisk and do a zap restart to clear out the channels and get it working again.. what would cause this, and how do i go about fixing it? |
| 16:39.05 | telnettech | i am looking at the phone settings and the "subscribe for MWI" is set to no |
| 16:44.48 | *** join/#asterisk zchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com) |
| 16:45.06 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:45.52 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 16:46.24 | telnettech | ET: I compared it to a working system and the Settings for both phones match |
| 16:49.01 | fcois93 | sobody to help me tonight... |
| 16:49.08 | fcois93 | have fun, bye! |
| 16:49.13 | etfonhomey | telnettech, I would try to catch the SUBSCRIBE message from 505 when it's booted up. Pastebin that (if it's sending one at all). |
| 16:50.09 | telnettech | ok so reboot the phone capture the sip messages. I should look for Subscribe |
| 16:50.26 | etfonhomey | Yep. |
| 16:50.59 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 16:51.18 | Aces1up | any ideas on my issue etfonhomey? |
| 16:52.49 | etfonhomey | Aces1up, no clue, but I'd like to know the answer when you find out. I'm getting ready to do an install using an A200 card. Does yours have hardware echo cancellation? |
| 16:53.18 | Aces1up | yes, but when i do a zap show channel x it says it is not enabled for some reason. |
| 16:53.40 | pecanha | hey.. when I put allow=ilbc&ulaw on trunk -> softphone works and ATA does not. When I put allow=ulaw&ilbc -> ata works and softphone does not. what the heck is this? |
| 16:56.19 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 16:56.44 | dandre | Hello, |
| 16:58.25 | dandre | I am willing to put a call on hold using dialplan. How could I do this ? How cna I unhold the call? Is there any tip for doing that? |
| 17:04.51 | telnettech | ET: after restarting the phone, the MWI is working now....interesting |
| 17:07.16 | *** join/#asterisk rpr_rpr (n=chatzill@73.Red-88-29-93.staticIP.rima-tde.net) |
| 17:07.19 | *** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 17:07.44 | rpr_rpr | Anyone knows if B410P needs an external power feed to actuate in NT mode? |
| 17:08.45 | rpr_rpr | because i'm trying to connect to a samsun PBX but L1 neves goes up. |
| 17:09.53 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:10.39 | kerx | anyone know of a valid US phone regex? |
| 17:12.31 | tzafrir_laptop | rpr_rpr, you need external power if you want to feed a phone that relies on external power |
| 17:12.38 | jer_ | kerx, NXXNXXNXXXX |
| 17:12.48 | jer_ | or you talking perl-style regex? |
| 17:12.59 | kerx | yeah :) |
| 17:13.03 | tzafrir_laptop | Otherwise (e.g. for a simple NT-TE loop between two ports of the card) you don't need it |
| 17:13.15 | telnettech | let me ask a basic question |
| 17:13.48 | jer_ | \d{3}\d{3}\d{4} |
| 17:13.52 | telnettech | i have a macro that works for all my internal numbers and takes care of checking channel status and going to voicemail and all that |
| 17:14.09 | telnettech | i have a line and it says this |
| 17:14.23 | telnettech | exten=>_[5]XX,1,Macro(stdexten-vm,${EXTEN}) |
| 17:14.48 | telnettech | so it is looking at the extension that the person is dialing and setting that as the variable for the macro |
| 17:15.46 | telnettech | instead of using ${EXTEN} variable, I can put an actual extension number in place of that and send that to the macro to use to follow the same call flow....is that correct? |
| 17:16.10 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 17:16.21 | telnettech | so i can do this.... |
| 17:16.21 | *** join/#asterisk asheron (n=stephan@78-27-56-63.dsl.alice.nl) |
| 17:16.49 | telnettech | exten=>4610,1,Macro(stdexten-vm,505) |
| 17:19.49 | etfonhomey | telnettech, I believe so. |
| 17:20.43 | etfonhomey | telnettech, just saw what you said about rebooting the phone. It must be running Windows. :) |
| 17:21.00 | etfonhomey | telnettech, I prever Polycom's. |
| 17:21.04 | etfonhomey | prefer* |
| 17:21.30 | telnettech | i can get into the web gui of the phone and check things and reboot if needed |
| 17:21.38 | telnettech | from a windows machine yes |
| 17:21.38 | *** join/#asterisk andrewy (i=andrewy@209.126.180.153) |
| 17:21.45 | telnettech | and that macro thing does work |
| 17:23.38 | *** join/#asterisk japerry (n=japerry@drupal.org/user/45640/view) |
| 17:24.56 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:27.05 | flujan | hello guys, I am using realtime queue members on my asterisk box. And i am having a strange behavior. Wanna check if it is a bug. |
| 17:27.14 | flujan | I have two agents on the queue. |
| 17:27.22 | flujan | Agent A answers a call. |
| 17:27.31 | flujan | Agent B answers a call. |
| 17:27.39 | flujan | A new call enters and wait in the queue. |
| 17:27.50 | dahunter3 | I read somewhere that an appropriate dial string will allow g729 to flow from end point to end point. What does that dial string option look like so that it will always try to do end point to end point instead of routing calls through asterisk? |
| 17:27.51 | flujan | I insert a new row on the queue_members table with AgentC |
| 17:28.10 | flujan | the call is not delivered to AgentC until a new call enters in the queue... |
| 17:28.16 | flujan | this is the expected behavior? |
| 17:31.31 | *** part/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 17:43.44 | *** join/#asterisk Pagautas (n=bigman@ns.voip.ktu.lt) |
| 17:50.35 | angryuser | hello i am trying to configure opensips-cp but siptrace allways reporting sorry -- cannot connect to xmlrpc server |
| 17:50.35 | angryuser | 111 - Connection refused |
| 17:51.22 | angryuser | i have installed xmlrcp module also enabled in php.ini , and my db configuration is fine, what am i doing wrong ? |
| 17:53.50 | angryuser | oh wring channel ;) |
| 17:53.50 | Katty | i am pissed off. |
| 17:53.55 | angryuser | wrong* |
| 17:54.00 | Katty | my company is going to sell toshiba systems |
| 17:54.03 | Katty | with 66 blocks |
| 17:54.05 | Katty | and analog phones |
| 17:54.09 | Katty | can you BELIEVE this nightmare? |
| 17:54.18 | Katty | and not just toshiba, they're selling samsung, and shortel and talkswitch |
| 17:54.20 | tzanger | Katty: that is teh awesoem |
| 17:54.45 | sprite-- | How do I make outgoing call to a cellphone with Teliax? I thought it was SIP/{$EXTEN}@teliax? |
| 17:54.50 | jaytee | oooooh! 66 blocks. they're almost as cool as bix blocks |
| 17:55.06 | Katty | i have the urge to just hand in my two weeks notice right now. |
| 17:55.13 | tzanger | hey, I *LIKE* Bix blocks |
| 17:55.29 | Katty | this company is so backwards. |
| 17:55.40 | Katty | but hey, it's southern missouri, what do i really expect |
| 17:55.48 | jaytee | Katty, AT&T is getting ready to lay off 4% of it's workforce. You might want to just stick with it and ride the storm out at this point. |
| 17:55.59 | Katty | sighs |
| 17:56.06 | tzanger | Katty: what's your company do anyway |
| 17:56.14 | Katty | tzanger: we're a copier dealer. |
| 17:56.15 | tzanger | besides sell toshiba systems with analog phones |
| 17:56.16 | jaytee | tzanger, and I liked 50's tv programming but I've moved on :-) |
| 17:56.23 | tzanger | jaytee: what's better than bix blocks? |
| 17:56.28 | tzanger | (I mean for interconnect) |
| 17:56.32 | Katty | tzanger: who decided to dabble in video surv. and sprint/nextel cellphones, and computer networks, and microsoft servers |
| 17:56.37 | Katty | tzanger: we specialize in NOTHING. |
| 17:56.40 | jaytee | standard 110 which is what replaced bix |
| 17:56.52 | Katty | tzanger: theritegroup.com |
| 17:56.54 | tzanger | meh |
| 17:57.03 | tzanger | 110 is just a nastier, slightly non-bix bix |
| 17:57.11 | tzanger | Bell Canada seems to be allbix |
| 17:57.14 | Katty | i don't get how this happened to me. |
| 17:57.21 | Katty | i was a microsoft server tech |
| 17:57.26 | Katty | who decided linux was nicer. |
| 17:57.26 | jaytee | or a better bix than bix depending on how you see things :-) |
| 17:57.27 | tzanger | Katty: heh |
| 17:57.30 | kaldemar | sprite--: depends totally on your configuration. it's not only the dial line that defines it. |
| 17:57.34 | Katty | and who decided to set up a phone system for the company that was /nice/ |
| 17:57.43 | Katty | and not pay several thousands of dollars |
| 17:57.57 | Katty | how did Katty the server tech get looped into running RJ45 cable |
| 17:58.02 | Katty | with Toshiba analog systems |
| 17:58.10 | jaytee | and then you found yourself strung out and being hammered by idiots? been there, done that, bought the whole inventory of tshirts |
| 17:58.15 | Katty | i work with my brain, NOT a punchdown tool |
| 17:59.27 | Katty | i have soft hands )= |
| 17:59.56 | Katty | is being turned into a janitor |
| 18:03.53 | *** join/#asterisk ziram19 (n=chatzill@196.203.52.254) |
| 18:04.02 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 18:06.35 | *** join/#asterisk bradleyprice86 (n=chatzill@fw.datafax.net) |
| 18:09.47 | telnettech | anybody ever heard of a PIX 25? i think they are mispronouncing it |
| 18:10.19 | Leddy | is there a max number of messages a user can have in their inbox? |
| 18:11.03 | telnettech | Leddy: it depends on what you set n your voicemail.conf and the HDD spce you have |
| 18:12.07 | telnettech | Leddy: there is a parameter that is maxmsg=xxx where xxx is the number of messages |
| 18:13.32 | Leddy | let me look at the conf |
| 18:14.16 | jaytee | does Asterisk 1.4 support h264 for video or just h263? |
| 18:15.29 | Leddy | maxmsg |
| 18:15.29 | Leddy | This limits the number of messages in a voicemail folder. The maximum value is 9999 (hard coded) and the default 100. When a mailbox has more than this number of messages in it, new messages can not be recorded and vm-mailboxfull is played to the caller. No more messages possible is also logged. This setting was introduced in 1.2. In previous versions it was hard coded to 100. |
| 18:15.34 | Leddy | thanks telnettech |
| 18:15.56 | telnettech | Leddy: np |
| 18:16.44 | *** join/#asterisk iratik (n=itariki@209.248.216.146.nw.nuvox.net) |
| 18:16.48 | iratik | What is the default root password? |
| 18:17.05 | iratik | I need to grant DB access to a client on another machine so i can create a "missed calls report" |
| 18:18.38 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:18.47 | tompaw | guys, my brain's overloaded. asterisk works with UDP, right? |
| 18:19.01 | iratik | rtp is a udp protocol i think |
| 18:19.26 | tompaw | thanks |
| 18:22.12 | jaytee | telnettech, Cisco makes a PIX 525 firewall |
| 18:22.56 | telnettech | jaytee: this is ome type of punch down block.....i think they mean Bix 25 |
| 18:23.09 | jaytee | ah |
| 18:24.16 | *** join/#asterisk linuxer_igor (n=linuxer_@mlsrj200152096p202.mls.com.br) |
| 18:25.50 | *** join/#asterisk farkus__ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:27.59 | telnettech | Im going to Aruba next week to do an install and the local cable vendor says they dont use 66 blocks |
| 18:33.05 | jaytee | telnettech, you can't use a 110 punchdown blade on a BIX block. |
| 18:33.14 | telnettech | I know that |
| 18:33.27 | jaytee | ok, ok!! just sayin! :-) |
| 18:33.43 | jaytee | at least you get to go to Aruba |
| 18:34.06 | telnettech | yeah......gotta love company paid "vacations" |
| 18:34.34 | jaytee | yeah, the last company paid vacation I went on was in Huntsville :-) |
| 18:35.27 | telnettech | mine was Lubbock, Tx |
| 18:36.00 | angryuser | i had more chance it was 'Ile de la reunion" (island ocean....very blue) |
| 18:37.09 | *** part/#asterisk Guest56564 (n=renemend@200.34.66.137) |
| 18:37.39 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:37.50 | angryuser | here it is http://ile-de-la-reunion.info/carte-la-reunion.html |
| 18:49.30 | *** join/#asterisk Pagautas (n=bigman@ns.voip.ktu.lt) |
| 18:52.23 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 19:01.46 | *** join/#asterisk Meaty (n=patate@office.abi.ca) |
| 19:05.25 | *** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 19:09.01 | *** join/#asterisk hfb (n=hfb@pool-96-229-38-185.lsanca.dsl-w.verizon.net) |
| 19:09.27 | *** join/#asterisk grx0 (n=dave@64-150-178-3.kansascity.abac.net) |
| 19:11.18 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 19:11.43 | ruben23 | hi |
| 19:13.58 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 19:14.21 | Meaty | Hi all, a little question for you. In dialplan (extension.conf), how can I call a macro with only one string argument containing a ",". "\" before "," was workin on 1.2, but isn't workin on 1.6 |
| 19:16.16 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 19:19.30 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 19:19.44 | lmadsen | Meaty: did you read the UPGRADE.txt and CHANGES file? It mentions something about that in those files I'm pretty sure. |
| 19:20.12 | DrAk0 | whats a good softphone for mac |
| 19:20.17 | DrAk0 | with g729 support |
| 19:20.20 | bradleyprice86 | x-lite |
| 19:20.34 | Meaty | lmadsen : no, its on my way. Thx. |
| 19:20.52 | lmadsen | Meaty: really you're supposed to read those files before upgrading, not after |
| 19:21.03 | lmadsen | DrAk0: eyeBeam |
| 19:22.50 | *** join/#asterisk sprite-- (n=sprite@12.228.1.131) |
| 19:22.56 | sprite-- | Anyone here use Teliax? |
| 19:23.59 | lmadsen | no, I use unlimitel |
| 19:24.59 | beek | jaytee: you around? |
| 19:26.29 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 19:27.29 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 19:40.19 | *** part/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 19:43.05 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 19:44.47 | beek | sprite--: Yes, I do. |
| 19:44.57 | *** join/#asterisk Segnale007 (n=Pietro@host88-253-dynamic.42-79-r.retail.telecomitalia.it) |
| 19:47.42 | jasonwoot | lmadsen: could NOT specifying localnet in SIP prevent listening on eth1? |
| 19:47.56 | lmadsen | does not know |
| 19:47.59 | lmadsen | you could always test? |
| 19:49.05 | jasonwoot | no worky in either case... it just won't seem to listen on two IPs on two NICs |
| 19:49.32 | lmadsen | I honestly just don't think that is possible. |
| 19:49.45 | jasonwoot | "24-96-139-164.knology.net.5060: UDP, length 498 |
| 19:50.00 | jasonwoot | er... |
| 19:50.01 | jasonwoot | "If omitted, the port will be set to 5060, and all |
| 19:50.01 | jasonwoot | IP addresses in your Asterisk system will accept incoming SIP connections. |
| 19:51.03 | *** join/#asterisk rene- (n=renemend@200.34.66.137) |
| 19:51.04 | rene- | hey |
| 19:52.28 | rene- | <PROTECTED> |
| 19:57.01 | *** join/#asterisk CorenWork (n=marc@2002:d133:c41a:0:0:0:0:c0de) |
| 19:58.28 | rene- | join #fop |
| 19:59.53 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 20:01.36 | CorenWork | Hey, I have a (possibly) simple question. Is it possible to have a caller execute stuff interactively in a context until a phone picks up that was dialed asynchronously? |
| 20:04.39 | lmadsen | CorenWork: possibly... I'm thinking some sort of combination of Local channels, Bridge() application, and maybe an AGI() or something like that... |
| 20:04.47 | *** join/#asterisk Segnale007 (n=Pietro@host88-253-dynamic.42-79-r.retail.telecomitalia.it) |
| 20:05.00 | kerx | -- Got SIP response 500 "Maximum Concurrent Connection Reached" back from |
| 20:05.00 | kerx | eew |
| 20:05.11 | lmadsen | I can't think of an easily, straight forward implementation, but I'm sure it is technically possible |
| 20:05.24 | kerx | congestion is terrible. |
| 20:05.53 | CorenWork | lmadsen: That's the direction I was looking at; but I hit my wall on the fact that dialing from an AGI disconnects the channel. Might have to be done with a custom app or through the API via originate. |
| 20:06.04 | lmadsen | aye |
| 20:06.37 | CorenWork | lmadsen: I was hoping someone would just already have solved that problem. :-) |
| 20:06.44 | lmadsen | CorenWork: doubt it |
| 20:06.51 | lmadsen | this room is kinda on the simple side :) |
| 20:06.58 | CorenWork | -dev? |
| 20:07.09 | lmadsen | CorenWork: no, that is not level 2 or 3.14 support |
| 20:07.25 | lmadsen | that is only for discussion of the asterisk code itself |
| 20:07.45 | CorenWork | Yeah, I expected as much. But chances are I'll have to write an app for it so I might as well make certain that it's done right so that others can use it. :-) |
| 20:08.04 | lmadsen | this is the appropriate venue, but you'll either need to hire a consultant to solve the problem, or work through it yourself, possibly asking specific questions with pastebin info |
| 20:08.30 | lmadsen | CorenWork: in that case you could use -dev if you're going to create a C module for asterisk |
| 20:08.39 | lmadsen | but not for external modules via AGI or AMI |
| 20:08.45 | lmadsen | that is still in this room |
| 20:08.53 | CorenWork | I've been looking at the dial app code, and the queue app code, which are the most similar. |
| 20:09.06 | Katty | bleh. |
| 20:09.10 | Katty | pouts, muchly |
| 20:09.25 | CorenWork | The problem is that I haven't found a good doc base for the internals. |
| 20:09.58 | lmadsen | CorenWork: have you looked at the doxygen docs? |
| 20:10.22 | lmadsen | CorenWork: http://www.asterisk.org/developers/documentation |
| 20:10.34 | CorenWork | Hm, nope. Looking now. |
| 20:10.37 | lmadsen | CorenWork: and http://www.asterisk.org/developers/ in general |
| 20:10.46 | lmadsen | lots of developer information there... funny enough :) |
| 20:10.52 | CorenWork | Search failiure for me not to have found this first. :-) |
| 20:11.00 | Katty | decorates lmadsen with ornaments. |
| 20:11.08 | lmadsen | glitters gleefully |
| 20:11.45 | CorenWork | lmadsen: I just wanted to see if there was a known easy way to do it before I start down the hard way. :-) |
| 20:11.49 | Katty | lmadsen: do you work with analog phone systems, and 66 blocks? |
| 20:11.59 | lmadsen | Katty: no, I'm smarter than that :) |
| 20:12.03 | grx0 | ouch |
| 20:12.05 | Katty | lmadsen: guess what my new jobis. |
| 20:12.14 | lmadsen | burn :) |
| 20:12.21 | Katty | lmadsen: they should change my title on my business card |
| 20:12.32 | Katty | lmadsen: Telcom-Janitor |
| 20:12.58 | Katty | lmadsen: needless to say, i'm not very happy about this whole thing. |
| 20:13.24 | rene- | i got tons of asterisk ami disconnects, what is manager writetimeout good for? |
| 20:14.29 | Katty | jbot: isymphony? |
| 20:14.30 | jbot | extra, extra, read all about it, isymphony is http://www.i9technologies.com/ |
| 20:14.47 | Katty | jbot: does it work with 1.6 yet? )= |
| 20:14.58 | seanbright | and more importantly - does it suck less? |
| 20:14.59 | grx0 | Katty: thats always fun....! Nothin is as much fun as re-doing someone elses poorly done work. |
| 20:15.20 | Katty | grx0: i don't think you quite understand. |
| 20:15.40 | Katty | grx0: my company is now PROMOTING old analog toshiba and samsung systems |
| 20:15.43 | Katty | grx0: because they are cheap |
| 20:15.55 | grx0 | wow, do they hate you? |
| 20:15.56 | Katty | grx0: and they have elected Katty, a server tech, to do analog phone crap. |
| 20:16.03 | grx0 | because it sounds to me....like they hate you. |
| 20:16.04 | *** join/#asterisk Segnale007 (n=Pietro@host88-253-dynamic.42-79-r.retail.telecomitalia.it) |
| 20:16.17 | Katty | grx0: they are doing it because people don't want to buy a good phone system. |
| 20:16.24 | Katty | grx0: they want something that will get them by for a few years |
| 20:16.28 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 20:16.38 | Katty | grx0: that is so cheap, you could practically buy two tin cans and some string for every office |
| 20:16.39 | CorenWork | Bridge() is 1.6, right? |
| 20:16.49 | Deeewayne | offers Katty a lollipop |
| 20:16.52 | grx0 | Katty: was that a marketing idea, a sales idea, or a mgmt idea? or a combination? |
| 20:16.53 | Katty | grx0: i seriously considered turning in my two weeks today. |
| 20:17.05 | Katty | Deeewayne: i think i'm going to need more than a lillipop |
| 20:17.23 | Katty | grx0: probably sales. all they care about is a price tag. |
| 20:17.26 | grx0 | hands Katty a benzo-pop. |
| 20:17.53 | grx0 | Katty: yeah, figures...why should they have to work harder at their job when they can just make you do the work instead? :) |
| 20:18.07 | Katty | more like clean up their mess. |
| 20:18.10 | Katty | they make the promises. |
| 20:18.19 | Katty | i make sure our company lives up to them. i bend over backwards for this place. |
| 20:18.40 | Katty | grx0: what makes me hate it even more, is i'm the ONLY person here that does this. |
| 20:18.50 | seanbright | holy crap |
| 20:18.50 | Katty | grx0: all the servers. |
| 20:18.53 | grx0 | yes, of course, thats the usual sales / technical dept. interaction at all companies. |
| 20:18.55 | seanbright | stop your whining |
| 20:18.56 | Katty | grx0: all the phone systems. |
| 20:18.57 | seanbright | heh |
| 20:19.00 | Katty | seanbright: i am a female. |
| 20:19.04 | Katty | seanbright: i will whine until i'm DEAD |
| 20:19.06 | seanbright | way to act like one |
| 20:19.14 | Katty | it makes me feel better ;) |
| 20:19.17 | seanbright | question: are you doing this work for free? |
| 20:19.25 | seanbright | question: are you unable to find other employment? |
| 20:19.36 | grx0 | Katty: look at the brightside, all that work with analog systems will for sure require a punch down tool, which given the right dark alleyway, could be used to inflict serious pain on some unsuspecting sales droid. |
| 20:19.49 | Katty | grx0: now that's the best idea i've heard all day :> |
| 20:20.25 | Katty | seanbright: and as for you, uncle scrooge. |
| 20:20.44 | grx0 | Katty: yeah, make sure its set on high and not that pansy 'gentle' setting... you know the one with the really really obnoxious loud pop... |
| 20:20.51 | Katty | decorates seanbright with red christmas bows |
| 20:20.55 | seanbright | Katty: no... i'm the public-pity-party-pooper |
| 20:21.05 | Katty | seanbright: someone always has to be. i'm used to it. |
| 20:21.09 | Katty | seanbright: won't keep me from bitching about it. |
| 20:21.21 | seanbright | that's fine |
| 20:21.25 | Katty | grx0: i've never actually used a punch down tool. |
| 20:21.28 | seanbright | just stating my obvious objection |
| 20:21.35 | Katty | seanbright: duly noted. |
| 20:21.37 | *** join/#asterisk daniev (n=dnv@190.144.60.154) |
| 20:22.00 | Katty | seanbright: i'll be fine in a few hours. |
| 20:22.14 | seanbright | splendid |
| 20:22.20 | Katty | i knew you'd approve. |
| 20:22.21 | seanbright | someone /msg me when that comes to pass |
| 20:22.26 | Katty | *hee* |
| 20:22.33 | grx0 | Katty: well you should, on your sales team, repeatedly. |
| 20:22.54 | Katty | grx0: ah well, they're just trying to keep the sales going |
| 20:23.19 | grx0 | well, then look as it as motivation for them to try harder...and stop selling legacy analog misery to customers. |
| 20:23.24 | Katty | they'd sell a cheeseburger to seanbright if they could make a buck |
| 20:24.04 | Katty | grx0: they don't think of it that way |
| 20:24.16 | Katty | grx0: they look at it as having a lot of 'options' and picking the one that fits the customer best. |
| 20:24.30 | Katty | grx0: and if that's a tin can with two strings, so be it :P |
| 20:24.54 | grx0 | Katty: analog pbx's aren't an option... unless you're into masochism, or really I guess it wouldnt be them inflicting misery on themselves, just you...lol. |
| 20:25.07 | *** join/#asterisk gpled (n=gpled@firewall.fccfurn.com) |
| 20:25.14 | Katty | well it's like seanbright said. |
| 20:25.25 | gpled | anyone know if an inexpensive wifi phone? |
| 20:25.30 | Katty | the company has already made the choice that they are goign to sell analog phone systems |
| 20:25.36 | Katty | i can shut up and do it, or turn in my two weeks |
| 20:25.41 | grx0 | gpled: aastara makes a decent one, I'd hardly call it inexpensive |
| 20:25.47 | lmadsen | the only wifi phone that seems to work well is the one built into my Nokia E61i |
| 20:25.53 | lmadsen | (in my experience) |
| 20:26.00 | grx0 | Katty: thats not as much fun as bitching about it for a couple hours and -then- getting over it |
| 20:26.08 | Katty | grx0: i know! |
| 20:26.14 | lmadsen | wifi sip phone + expensive = divide by zero |
| 20:26.17 | Katty | grx0: must be a girl thing :P |
| 20:26.26 | lmadsen | s/expensive/inexpensive/ |
| 20:26.30 | Katty | seanbright: k, i'm sufficiently over it. |
| 20:26.34 | Katty | seanbright: you can come out of hiding. |
| 20:26.45 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 20:27.20 | grx0 | the aastara 9480i works awesome on my system |
| 20:27.32 | grx0 | thats not wifi tho |
| 20:27.47 | grx0 | but wireless/desktop combo, works alot better than any wifi ones I tried...they were all crap |
| 20:28.08 | Katty | plantronics? |
| 20:28.11 | grx0 | Katty: well I'm not a girl, but I do enjoy bitching about things...so..whatever :D |
| 20:28.39 | *** join/#asterisk azamzia (n=azam_zia@h-79-136-91-158.NA.cust.bahnhof.se) |
| 20:28.44 | Katty | feels good to get the rage out ^_^ |
| 20:29.09 | lmadsen | rages against the machine |
| 20:29.26 | Katty | offers lmadsen punch-down tool |
| 20:29.37 | lmadsen | have one, thanks! :) |
| 20:30.02 | grx0 | lol yess the punchdown tool, the appropriate way to send a 'message' to the sales droids, when used correctly at least. |
| 20:30.27 | lmadsen | make a desk, finger, punchdown tool sandwich! make sure you press down |
| 20:30.39 | azamzia | Hi Can anyone please tell me what is this? Disconnecting call 'SIP/192.168.1.1-b26f7928' for lack of RTP activity in 513 seconds |
| 20:30.44 | Katty | lmadsen: do you cut yourself much with those? |
| 20:30.44 | gpled | grx0: lol |
| 20:30.57 | grx0 | yeah, ow, I'm not sure if that would result in just a bone contusion or a broken finger...but I know it would hurt like (@&% |
| 20:31.00 | Corydon76-dig | I thought that was a handcuff-lmadsen-to-the-bed tool |
| 20:31.03 | lmadsen | azamzia: means no RTP activity was detected and the call was disconnected |
| 20:31.14 | lmadsen | Corydon76-dig: you are wrong sir |
| 20:31.38 | lmadsen | Katty: have never cut myself with a punchdown tool (lucky?) |
| 20:31.43 | azamzia | @lmadsen: y m i getting this mesg? and how to stop this? |
| 20:31.48 | Corydon76-dig | Huh, I wonder what this tool is for, then.. |
| 20:31.57 | Katty | lmadsen: not sure, to be honest. |
| 20:31.59 | grx0 | lmadsen: itd take some talent to cut yourself with a punchdown tool on accident... |
| 20:32.02 | gpled | lmadsen: you'v learned to keep you finger away from the end too :) |
| 20:32.03 | grx0 | or on purpose for that matter. |
| 20:32.08 | Katty | lmadsen: every time i end up running cable, i manage to really make my fingers sore. |
| 20:32.22 | Katty | lmadsen: for a few days later |
| 20:32.24 | lmadsen | azamzia: it means asterisk has no detected any audio/media/RTP activity for a period of time and disconnected the call. See sip.conf.sample for more information about that option. |
| 20:33.00 | lmadsen | Katty: oh ya, me too, but that more has to do with running cable through my fingers, and nothing to do with the punchdown tool |
| 20:33.37 | Katty | cabling is torture. |
| 20:33.39 | grx0 | Katty: How much is a fine samsung analog box of hate running these days anyway according to your sales people? |
| 20:33.47 | Katty | grx0: i've no idea. |
| 20:33.52 | Katty | grx0: i'm trying to stay out of that area. |
| 20:34.02 | azamzia | @lmadsen: Thank You, i saw sip.conf while trying to debug this myself but my question is if hangup has already been requested on the channel then still y m i getting the NOTICE and also y isnt the channeld disconnected even after 513 seconds |
| 20:34.07 | *** part/#asterisk gpled (n=gpled@firewall.fccfurn.com) |
| 20:34.21 | azamzia | @lmadsen: i m getting lots of these msgs and i m not sure what to do with them .... |
| 20:34.32 | fogo | I've got a PRI that keeps bouncing. Can anyone take a look at this intense debug and tell me what's going on: http://pastebin.com/d361bff3d |
| 20:34.44 | grx0 | Katty: what wisdom you exhibit! |
| 20:35.29 | jasonwoot | "kill_zombies: This option should not be used, it is disabled by default (0). " George Romero would be pissed |
| 20:36.00 | grx0 | fogo: what signaling is that supposed to be? |
| 20:36.22 | fogo | grx0: pri_net |
| 20:36.34 | lmadsen | azamzia: sounds like the call is not being disconnected correctly |
| 20:36.43 | Katty | grx0: i'm just hoping sanity comes to them, ya know? |
| 20:36.52 | Katty | grx0: they'll look at the pricing, and what they get, and it wil ljust somehow click |
| 20:37.43 | azamzia | @lmadsen: can u please suggest any reasons for that or what should i do with it? can i reduce some timeout or is there a way to stop this message? cant we forcefully hangup a chennel? |
| 20:38.04 | grx0 | Katty: was that supposed to be funny? I found it rather amusing...lol |
| 20:38.09 | lmadsen | azamzia: you need to debug why the channel is not hanging up correctly. Most likely it is because the appropriate SIP messages are not being received. |
| 20:38.25 | Katty | grx0: ;) |
| 20:38.47 | grx0 | fogo: what protocol is it, and you're providing the network side? to a pbx or something? |
| 20:39.10 | azamzia | @lmadsen: ok i will debug the sip traffic, but is there way to stop these messages or forcefully hangup these channels? |
| 20:39.31 | azamzia | @lmadsen: i tried soft hangup but no success |
| 20:39.40 | grx0 | Katty: I wish I had more faith in peoples ability to not screw stuff up on a monumental level, especially when it comes to technology. lol. |
| 20:40.03 | lmadsen | azamzia: not to my knowledge, you need to debug the underlying issue, or lower the rtp disconnect value (i forget the exact name of the option in sip.conf) |
| 20:41.15 | Katty | grx0: i know the feeling. |
| 20:41.32 | Katty | grx0: not a whole lot lil ole me can do about it tho |
| 20:41.51 | Katty | grx0: except when they screw up, tell them i told you so ;) |
| 20:42.03 | fogo | grx0: we're providing the network side to the telco (I thought that was odd, but that's the way they wanted it). zapata: http://pastebin.com/d124c754a zaptel: http://pastebin.com/db26d9f7 |
| 20:42.51 | azamzia | @lmadsen: its rtptimeout, rtpholdtimeout i have already set them to 20seconds |
| 20:43.48 | grx0 | fogo: wow....that -is- odd.. especially one just one span... |
| 20:44.23 | fogo | grx0: all our others are cpe.. just this one they said -we- are net (Qwest) |
| 20:44.25 | grx0 | Katty: yeah thats usually at least a little gratifying, getting to tell sales/marketing I told you so... |
| 20:44.27 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 20:44.45 | grx0 | fogo: tell them to stop smoking the crack and get a technician that can configure a PRI span correctly on their hardware lol. |
| 20:44.46 | lmadsen | azamzia: sorry, I have no additional opinions at this time |
| 20:45.03 | fogo | grx0: lol |
| 20:45.23 | grx0 | fogo: thats just my opinion, I dunno from the looks of it, but I'd demand them to put my span in a 'normal' or at least 'semi-normal' configuration before I'd even waste time troubleshooting it if it hasn't worked before. |
| 20:45.40 | grx0 | now if it has worked before I'm an idiot and need to shut my mouth, but I'd assume it hasnt from what you've told me thus far? |
| 20:46.01 | azamzia | tly on their hardware lol. |
| 20:46.01 | azamzia | @lmadsen: Thanks alot :) |
| 20:47.03 | fogo | grx0: we just got it turned up today. Took me awhile to figure out why it wasn't working - then he told me we were net... it comes up, I can make a call out (when it's up), but it'll die when it bounces |
| 20:47.30 | grx0 | fogo: when I've worked for 2 CLECs not once did we put a PRI in place with the customer as net, thats just....bizzarre. are they giving you guff about getting it changed? |
| 20:48.05 | fogo | grx0: I asked him to change us to cpe - he said they *normally* have the customer as net |
| 20:48.32 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
| 20:49.25 | *** join/#asterisk almir (n=almir@91.191.20.108) |
| 20:49.58 | fogo | grx0: I'll go harrass them later and see if they'll switch us to cpe.. I gotta run for now though. Thanks for the help |
| 20:50.00 | grx0 | fogo: he *normally* doesn't know what he's doing then IMHO, the only time I've had customer equipment as NET is if its connecting a tie line to other customer equipment.. I may not know everything, but I've turned up a few pri in my day...lol |
| 20:50.43 | grx0 | fogo: good idea ni2/cpe is pretty standard across the 16 or so states I've installed in so lol.... good luck! |
| 20:51.24 | fogo | grx0: yeah, there's nothing special here. just a standard PRI, like our other 4. I have no idea why they want go do it that way - I thought pretty much everyone did cpe *shrug* |
| 20:52.03 | grx0 | to be as difficult as they can, because they're the phone company, thats the luxury of being the phone company, doing stupid things and not caring what the customer thinks because it'd take em a month to switch carriers :) |
| 20:52.12 | fogo | lol |
| 20:52.20 | grx0 | thats if there even is other carriers in your area even lol |
| 20:53.20 | fogo | there are. we have at least one other line from another carrier; and at least one more from them (that -is- cpe) |
| 20:53.35 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 20:54.49 | grx0 | yeah I think you just got a retarded tech is all...I've dealt with qwest a few times and never had them request customer equipment as NET. hope all goes well for ya. |
| 20:55.08 | fogo | grx0: thanks :) |
| 20:58.02 | seanbright | pats Katty on the cabeza |
| 20:58.04 | seanbright | there there |
| 20:59.53 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 21:00.56 | unpaidbill | are all the fixes for chan_skinny in 1.6 applied to the 1.4 tree as well? |
| 21:03.46 | lmadsen | no |
| 21:03.51 | lmadsen | well... where relative |
| 21:04.06 | lmadsen | bug fixes go: 1.4 --> trunk --> 1.6.x |
| 21:07.23 | *** part/#asterisk CorenWork (n=marc@2002:d133:c41a:0:0:0:0:c0de) |
| 21:11.59 | legis | ~ITSP |
| 21:12.00 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 21:12.16 | legis | ~itsplist-us |
| 21:12.17 | 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 |
| 21:13.28 | *** join/#asterisk psofa (n=psofa@adsl43-160.ath.forthnet.gr) |
| 21:15.08 | *** join/#asterisk trumee (n=trumee@cpc1-nmkt1-0-0-cust91.cmbg.cable.ntl.com) |
| 21:15.17 | psofa | hello, i know the right solution would be a digium card but id like to experiment first.I have an isdn NT which can be configured like 'voice transparent' which means everything i send/receive from it goes to the B-channel.Communication with it is possible via a tty device.Is there any chance it might work with asterisk? |
| 21:15.20 | trumee | does anybody use nokia phone for sip? |
| 21:15.39 | unpaidbill | lmadsen cool thanks |
| 21:16.24 | unpaidbill | hopefully this means my 7935 wont lock up in 1.4.22 now, seems to work in 1.6.0.1 without a hitch |
| 21:16.50 | trumee | i was wondering whether it is possible to define an access point priority list on Nokia phones |
| 21:17.02 | trumee | The sip works great on wifi, but not so good on 3g |
| 21:17.09 | trumee | any nokia phone users here? |
| 21:17.48 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.66) |
| 21:20.16 | unpaidbill | nope, i'm waiting for nokia to make an awesome android phone, and waiting for android to suck less |
| 21:24.37 | trumee | unpaidbill, i dont think that will ever happen |
| 21:25.30 | unpaidbill | we'll know in 5 years |
| 21:26.41 | *** join/#asterisk farkus (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 21:31.17 | *** join/#asterisk docelmo (n=chatzill@206.248.239.194) |
| 21:31.36 | docelmo | Say anyone in here from Virginia? or within 3 hours drive of Roanoke, VA? |
| 21:34.57 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 21:37.34 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 21:39.31 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 21:43.06 | psofa | ill rephrase in case someone knows.I have an isdn NT which can be configured like 'voice transparent' which means everything i send/receive from it goes to the B-channel.Communication with it is possible via a tty device.Is there any chance it might work with misdn? i suppose what im looking for is layer 1 over serial port? |
| 21:43.34 | trumee | is it possible to have realm=blah on per extension basis? |
| 22:11.10 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 22:14.11 | kerx | what is better instead of using System() for asterisk optimisation? |
| 22:14.48 | [TK]D-Fender | kerx: What does "System()" have to do with "optimization"? |
| 22:15.14 | kerx | [TK]D-Fender, it seems like because I use System() to do logging on actions, Asterisk is extremely bottle necked |
| 22:15.37 | lmadsen | kerx: that's because you're instantiating an instance of bash everytime |
| 22:15.44 | lmadsen | kerx: use FastAGI() |
| 22:15.47 | kerx | yea :-( somebody told me about it before |
| 22:15.53 | kerx | oh ok... FastAGI() |
| 22:15.54 | lmadsen | you're using the wrong feature |
| 22:16.16 | kerx | FastAgi, let me check it out |
| 22:16.20 | kerx | my asterisk is at 90% |
| 22:16.22 | kerx | i'm so mad |
| 22:16.26 | kerx | grr... |
| 22:16.27 | kerx | heh |
| 22:18.00 | *** join/#asterisk JunK-Y (n=junky@modemcable156.137-20-96.mc.videotron.ca) |
| 22:26.01 | Katty | JunK-Y: OHAI |
| 22:26.28 | JunK-Y | ola! |
| 22:26.30 | JunK-Y | whats up? |
| 22:26.45 | Katty | you don't want to know )= |
| 22:26.46 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 22:26.51 | Katty | it will end in me ranting and raving for another hour. |
| 22:27.00 | Katty | how're you, dear? |
| 22:28.34 | JunK-Y | nothing, end of classes tomorrow. |
| 22:28.58 | JunK-Y | ive to jet for an hour. bbk |
| 22:31.26 | [TK]D-Fender | whispers to name "Samsung" |
| 22:31.33 | Katty | glares at [TK]D-Fender |
| 22:31.46 | [TK]D-Fender | lowers his visor |
| 22:31.48 | Katty | shakes punchdown tool in [TK]D-Fender's general direction |
| 22:32.25 | [TK]D-Fender | shakes a bix connector block in Katty's general direction |
| 22:32.50 | [TK]D-Fender | Katty: Meet me have way, we do wiring ;) |
| 22:32.55 | [TK]D-Fender | half* |
| 22:33.01 | [TK]D-Fender | bleh... barin = toast |
| 22:33.04 | [TK]D-Fender | brain! |
| 22:33.06 | [TK]D-Fender | ahdklasfhosdfyhiysd |
| 22:33.09 | [TK]D-Fender | convulses |
| 22:33.15 | Katty | twitches |
| 22:33.19 | Katty | do what? |
| 22:33.24 | [TK]D-Fender | wiring :) |
| 22:33.34 | Katty | do not., |
| 22:33.46 | Katty | [TK]D-Fender: i do my nails tho. |
| 22:33.54 | [TK]D-Fender | Katty: Silly programmer... you need to play with some hardware! |
| 22:33.54 | Qwell | sounds like a fair compromise |
| 22:33.58 | Katty | [TK]D-Fender: :< |
| 22:34.00 | Katty | [TK]D-Fender: no. |
| 22:34.03 | Katty | [TK]D-Fender: hardware is evil. |
| 22:34.16 | [TK]D-Fender | Katty: Besides.... with all this talk about "hard" and "floppy".. shouldn't we be having more fun? ;) |
| 22:34.18 | Qwell | Katty: 80. go. |
| 22:34.37 | Katty | Qwell: i have a nail polish color that's 80 miles from anywhere. |
| 22:34.53 | Katty | Qwell: it's a pretty lime green type color. |
| 22:35.27 | Katty | [TK]D-Fender: getting nicks and cuts on my hands is not pleasant. |
| 22:36.08 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.66) |
| 22:50.09 | *** join/#asterisk Firass-VC22 (n=firass@rza.vikcomm.wwu.edu) |
| 22:55.51 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-b133150c8aeea363) |
| 22:55.52 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 23:08.34 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 23:11.50 | harry_v | Im getting a error in cli that says "no application DigitTimeout for extention (contextname,extention,priority) the pastebin is at http://www.pastebin.ca/1276356 the extention is a working extention. This is just a goto statment after giving the caller a choice to call my extention or conferance. |
| 23:13.16 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 23:18.47 | *** part/#asterisk JBenden76_ (n=JBenden7@cblmdm72-240-122-140.buckeyecom.net) |
| 23:19.55 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:20.52 | JunK-Y | harry_v: simple, DIgitTimeout has changed to use the function TIMEOUT |
| 23:34.54 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-2f8d81f2b8350d23) |
| 23:34.54 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 23:34.56 | *** join/#asterisk Akiyuki (i=TuxGuy@cpe-065-184-194-136.ec.res.rr.com) |
| 23:37.18 | *** join/#asterisk harry_v (n=pcsuppor@S010600a0c93f6f7e.vs.shawcable.net) |
| 23:47.02 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 23:50.50 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 23:57.51 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 23:59.33 | *** join/#asterisk chodem (n=chodem@rrcs-24-173-37-76.sw.biz.rr.com) |
| 23:59.37 | *** part/#asterisk chodem (n=chodem@rrcs-24-173-37-76.sw.biz.rr.com) |