| 00:05.55 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
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| 00:15.14 | `paul | hi my originatecall takes a long time what could be the reason behind this? web server is on a separate machine (with the asterisk server) if i transfer the script(php) on the same machine of the asterisk server you guys think it will improve the waiting time? |
| 00:25.55 | *** join/#asterisk xpot-mobile (n=james@144.35.25.181) |
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| 01:07.27 | Zuchmir2 | ok, looks like asterisk -rx "convert foo.wav foo.g729" jumps out as before the convertion finishes... is there a way to have it wait until it's finished (so that it can be done from a shell script in batch)? |
| 01:09.59 | *** join/#asterisk tobias (n=tobias@user-0ce2hp1.cable.mindspring.com) |
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| 01:36.53 | *** part/#asterisk rene- (n=renemend@200.34.66.137) |
| 01:41.47 | ManxPower | Zuchmir2: My only suggestions are to make sure you have the latest of whatever branch of Asterisk you are using, if it's not fixed in that version, then file a report on bugs.digium.com. The asterisk -rx stuff has historically had problems getting all the output back to your shell. |
| 01:45.39 | Zuchmir2 | ManxPower: i basically added a while grep on show license, and wait for license to be available :-( |
| 01:46.46 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
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| 01:47.01 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 01:56.01 | lasko | Is there any documentation on using the schedule=yes in meetme.conf on the latest 1.6? |
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| 02:29.49 | tengulre | hi,all |
| 02:32.26 | *** join/#asterisk Wayhigh (i=wayhigh@www.kevinlynn.com) |
| 02:33.45 | Wayhigh | so I'm in the process of doing an LNP.. (it hasn't happened yet).. and for the last several days my asterisk box can see a ring.. answers it.. tries to play a message but the caller never hears anything but ring tones.. anyone seen this before? |
| 02:38.06 | ManxPower | Wayhigh: Yes. |
| 02:38.23 | ManxPower | But I think you'll need to provide some details like the tech the call is coming in on, etc. |
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| 02:40.34 | *** part/#asterisk yoblooc (n=yoblooc@210.83.214.163) |
| 02:48.56 | Wayhigh | manx: it's coming in on a tdm400p fxo |
| 02:49.27 | Wayhigh | the fxo has no problem doing outbound dialing |
| 02:49.39 | ManxPower | Wayhigh: have you tried power cycling the machine? |
| 02:49.48 | Wayhigh | manx: yeah |
| 02:49.55 | ManxPower | (that's what always solved my TDM400P problems) |
| 02:50.09 | Wayhigh | I'm thinking about changing the pci slot of the tdm |
| 02:50.17 | Wayhigh | I have an irq conflict I gotta resolve anyways |
| 02:50.21 | ManxPower | then the only thing I can think of is a line issue. Make sure there is no crossed lines on the punchdown block in the telco closet |
| 02:50.48 | ManxPower | I've seen similar problem when I had green from one line with red from another line going into the same port |
| 02:51.00 | ManxPower | i.e. screwed up wiring and/or punchdown |
| 02:51.00 | Wayhigh | hmm.. that's an interesting idea |
| 02:51.07 | Wayhigh | i'll have to check that out |
| 02:51.33 | *** join/#asterisk Selveste1 (n=Selveste@4103ds1-nks.0.fullrate.dk) |
| 02:51.40 | Wayhigh | I basically made a long extension cable that goes from a jack to my server closet.. I'm not punching down inside the closet.. it goes straight to the card |
| 02:53.38 | carrar | HOT++ |
| 02:53.57 | Wayhigh | OK |
| 02:54.53 | Wayhigh | carrar: you remember the days you could +++ATH0 and watch people on irc drop like flies? |
| 02:54.58 | carrar | haha |
| 02:55.04 | carrar | connectect.com |
| 02:55.07 | carrar | err |
| 02:55.11 | carrar | connected.com |
| 02:55.18 | Wayhigh | werd.. those were fun days |
| 02:55.24 | carrar | 89 was a fun year |
| 02:55.31 | Wayhigh | heh. osek's still not over me hitting him with flash.c |
| 02:56.29 | Wayhigh | hard to believe that was 20 years ago.. |
| 02:56.55 | carrar | wayhigh, does that card have a power port that needs to plug into your power supply? |
| 02:57.02 | carrar | I don't remember |
| 02:57.18 | ManxPower | carrar: that is only required for FXS ports |
| 02:57.23 | carrar | ah |
| 02:57.29 | carrar | thats a common mistake |
| 02:57.47 | ManxPower | The PCI bus on some systems can't handle the power draw of ringing all the ports |
| 02:57.56 | ManxPower | So the card uses external power. |
| 02:58.01 | Wayhigh | well the power is on the card itself but it's not used by anything but fxs ports I think |
| 02:58.17 | carrar | Wayhigh, get some audiocode FXO cards |
| 02:58.25 | Wayhigh | carrar: what're those? |
| 02:58.33 | carrar | well not cards |
| 02:58.38 | carrar | SIP to FXO |
| 02:58.52 | Wayhigh | I've been thinking about getting those usb fxo's from sangoma |
| 02:58.59 | carrar | ack |
| 02:59.19 | Wayhigh | I've heard rumor that there's a firmware update coming out for them soon |
| 02:59.53 | carrar | wayhigh: http://www.audiocodes.com/products/mediapack-1xx |
| 03:00.40 | Wayhigh | why're you acking about the usb fxo's? |
| 03:00.57 | carrar | not a fan of USB |
| 03:01.07 | Jumpie | Wayhigh lol |
| 03:01.09 | Jumpie | i remember that |
| 03:01.30 | Wayhigh | jumpie: you've been around that long too? |
| 03:01.35 | Jumpie | yup |
| 03:01.38 | Jumpie | i been irc'in since 94 |
| 03:01.41 | carrar | wayhigh, what are you doing that you need FXO? |
| 03:01.48 | Wayhigh | carrar: pstn |
| 03:01.52 | *** join/#asterisk tobias (n=tobias@user-0ce2hp1.cable.mindspring.com) |
| 03:01.53 | carrar | athome? |
| 03:01.57 | carrar | or work |
| 03:02.00 | carrar | for biz? |
| 03:02.16 | Wayhigh | I use a pstn line as a backup for my wife incase there's a massive power failure while she's working |
| 03:02.24 | Wayhigh | and for outgoing faxes |
| 03:02.28 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 03:02.28 | Wayhigh | it's just easier ya know |
| 03:02.34 | carrar | yeah |
| 03:02.50 | Wayhigh | brb.. need food |
| 03:06.38 | *** join/#asterisk BeeBuu (n=beebuu@125.95.250.223) |
| 03:07.50 | BeeBuu | any one tried conference manager from asterikast? |
| 03:13.42 | *** join/#asterisk ctp (n=ctp@brsg-d9bef7bc.pool.mediaWays.net) |
| 03:15.18 | Wayhigh | never even heard of asterikast |
| 03:15.31 | Wayhigh | anyone know of a cheap cdma gateway? |
| 03:16.12 | carrar | Wayhigh, you should go to astericon this year |
| 03:16.16 | carrar | I'll be there |
| 03:16.28 | carrar | we can hang out and throwback a few |
| 03:16.57 | [TK]D-Fender | Plenty of throwbacks here already ;) |
| 03:17.02 | carrar | hahah |
| 03:17.22 | carrar | I think I have seen a cell to SIP device someplace |
| 03:17.41 | carrar | that what you are looking for? |
| 03:18.21 | Wayhigh | carrar: where's astericon? |
| 03:18.29 | carrar | AZ |
| 03:18.31 | carrar | Glendale |
| 03:18.32 | Wayhigh | I was supposed to go to defcon but it looks like I may not be allowed to go now |
| 03:18.45 | carrar | Just north of PHX |
| 03:18.50 | Wayhigh | either that.. or I may not have a job by the time it comes around |
| 03:18.59 | carrar | heh |
| 03:19.02 | carrar | ouch |
| 03:19.16 | carrar | time to move back to your roots! |
| 03:19.27 | Wayhigh | naw.. I'm just going through some stuff |
| 03:19.28 | carrar | ice creame truck deliver her we go! |
| 03:19.31 | carrar | heh |
| 03:19.35 | Wayhigh | hahaha |
| 03:19.39 | Wayhigh | dude that job was fun.. |
| 03:19.42 | carrar | heh |
| 03:19.50 | Wayhigh | deliver icecream in the day.. haxor all night long |
| 03:20.42 | carrar | http://cgi.ebay.com.my/GP-712-WCDMA-CDMA-3G-SIP-VoIP-Gateway-2-voice-channels_W0QQitemZ110337737428QQihZ001QQcategoryZ61839QQcmdZViewItem |
| 03:20.49 | carrar | <PROTECTED> |
| 03:23.54 | ManxPower | The 3G Voice Revolution! |
| 03:24.12 | ManxPower | 3G voice is just so much better than that old 1G or 2G voice. |
| 03:24.23 | carrar | hahah |
| 03:24.36 | Wayhigh | carrar: dude.. that auction would be great.. if I were to live in Malaysia.. |
| 03:24.42 | carrar | well |
| 03:24.52 | carrar | was ment as a YES |
| 03:25.11 | carrar | letme know if I can google anything else for ya :) |
| 03:25.16 | carrar | heh |
| 03:25.22 | drmessano | http://cgi.ebay.com/GP-712-WCDMA-CDMA-3G-SIP-VoIP-Gateway-2-voice-channels_W0QQitemZ110337737428QQihZ001QQcategoryZ61839QQcmdZViewItem |
| 03:25.28 | drmessano | or just fix the domain name |
| 03:25.46 | ManxPower | Exactly how is 3G voice different from revular voice? |
| 03:25.50 | Wayhigh | heh.. free baja fish tacos from long john silvers.. |
| 03:25.50 | ManxPower | and regular voice too. |
| 03:25.57 | drmessano | Its not 3G voice |
| 03:26.11 | Wayhigh | that's just wrong man.. in so many ways.. everyone knows there's hardly any hot chicks at long john silvers |
| 03:26.17 | ManxPower | So it's a SIP data device? |
| 03:26.27 | carrar | wayhigh, it's only for 7 days |
| 03:26.32 | carrar | err |
| 03:26.34 | carrar | in 7 days |
| 03:26.36 | ManxPower | Yes, I know it's just a stupid attempt at using keywords to get more traffic, but it still irritates me. |
| 03:27.21 | ManxPower | Come to think of it most marketing irritates me. |
| 03:27.45 | drmessano | s/marketing/everything/ ? |
| 03:28.14 | ManxPower | drmessano: just stupidity in general |
| 03:28.47 | ManxPower | lying irritates me too, but I think that's covered under "marketing" |
| 03:34.38 | oilinki | drmessano: you are from/living in malaysia? |
| 03:35.22 | carrar | he is |
| 03:35.35 | carrar | he has a harim |
| 03:35.55 | carrar | karem |
| 03:35.57 | carrar | err |
| 03:35.59 | carrar | harem |
| 03:36.03 | oilinki | hih |
| 03:36.09 | *** join/#asterisk MoreAllLess (n=Justo@cpe-76-169-252-172.socal.res.rr.com) |
| 03:38.45 | oilinki | good to hear that some people are around here |
| 03:42.07 | BeeBuu | any one tried conference manager from asterikast? |
| 03:45.25 | oilinki | btw. are you able to use voip over EDGE? |
| 03:48.13 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 03:58.06 | [TK]D-Fender | BeeBuu: About as many as a half an hour ago |
| 04:01.30 | *** join/#asterisk MadMoney (n=Madmoney@cpe-24-93-138-183.maine.res.rr.com) |
| 04:01.34 | MadMoney | !savemoney |
| 04:01.47 | MadMoney | Thanks for deleting that factoid. |
| 04:02.05 | MadMoney | I'm working on building my own ATAs. |
| 04:02.29 | MadMoney | I call them acoustic coupler ATAs. |
| 04:03.00 | [TK]D-Fender | ~savemoney |
| 04:03.00 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 04:03.03 | [TK]D-Fender | :p |
| 04:03.11 | *** join/#asterisk xpot-mobile (n=james@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 04:03.33 | MadMoney | =( |
| 04:03.47 | MadMoney | I am saving money by making my own ATAs. |
| 04:04.09 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 04:04.20 | MadMoney | I am making them with sound cards and RJ11 telephones. |
| 04:04.32 | carrar | Your not the first! |
| 04:04.35 | MadMoney | Acoustic coupler. |
| 04:04.47 | carrar | Old Atari 830 modems? |
| 04:04.55 | carrar | those are HOT++ |
| 04:05.04 | MadMoney | Microphone to telephone speaker, speaker to telephone receiver (and them to USB sound card) |
| 04:05.25 | MadMoney | Basically a makeshift acoustic coupler. |
| 04:05.38 | carrar | You should write a white paper |
| 04:05.38 | MadMoney | If it's good enough for a TTY modem, it is good enough for a business. |
| 04:05.59 | carrar | take lots of pics |
| 04:06.04 | MadMoney | Why? |
| 04:06.04 | *** join/#asterisk ovrstorm (n=ovrstorm@host-69-145-242-34.grf-mt.client.bresnan.net) |
| 04:08.08 | [TK]D-Fender | madyGoing speaker to mic is RETARDED. just F-ing WIRE THEM TOGETHER. |
| 04:08.48 | [TK]D-Fender | MadMoney: You are converting from electrical to mechanical unnecessarily. |
| 04:08.48 | MadMoney | That's hard to do when the voltages are completely different. :) |
| 04:09.21 | [TK]D-Fender | MadMoney: Seriously You think you can make an ATA yourself that isn't utter garbage, and the drivers to support it, for less that 25$ per port? |
| 04:09.27 | MadMoney | 90 volts phone line wired to soundcard microphone in? :D |
| 04:09.42 | MadMoney | ATAs are $40. |
| 04:09.59 | [TK]D-Fender | MadMoney: Yeah, thats what TRANSFORMERS are for. That and giving us Megan Fox draped over "whatever" |
| 04:10.11 | WindowsUser | is there any free TTY software? I just want to play around with TTY for a day |
| 04:10.16 | [TK]D-Fender | madYou can get a **2-Port** ATA for 50$ or less <- |
| 04:10.39 | MadMoney | WindowsUser: telnet.exe |
| 04:10.39 | Nugget | telnet is eeeeeeevil! |
| 04:10.49 | MadMoney | telnet.exe |
| 04:11.20 | WindowsUser | well i meant like the TTY tone thing that deaf people use |
| 04:11.26 | *** part/#asterisk hesco (n=hesco@24.99.160.121) |
| 04:12.02 | [TK]D-Fender | MadMoney: http://www.telephonydepot.com/Catalog/Grandstream-Analog-Adapters/Grandstream-HandyTone-496-HT496 |
| 04:12.34 | [TK]D-Fender | MadMoney: 2 port < $20 per port |
| 04:12.43 | [TK]D-Fender | MadMoney: Your plan is a prawling mess |
| 04:12.47 | [TK]D-Fender | sprawling* |
| 04:13.03 | carrar | mmmm Megan Fox |
| 04:13.20 | [TK]D-Fender | MadMoney: Doing this as anything other than a "because I can" is absoutely retarded. |
| 04:13.44 | carrar | because he doen't know he can't |
| 04:13.48 | [TK]D-Fender | carrknew I'd get your attention ;) |
| 04:14.05 | WindowsUser | well dont most atas suck? |
| 04:14.20 | [TK]D-Fender | WindowsUser: No. |
| 04:15.55 | MadMoney | calls off DIY ATA project! |
| 04:16.03 | MadMoney | In the name of saving money. |
| 04:17.11 | [TK]D-Fender | MadMoney: Oh, and you need your sound-card plugged into a PC too. What about software? How are you going to pickup/hangup? What about hook-flash? Ring detection? How about acoustic isolation since you're COUPLING this psycho Franken-phone? |
| 04:17.36 | [TK]D-Fender | MadMoney: Oh... and then what software controls all of it? |
| 04:18.15 | carrar | xmodem! |
| 04:18.21 | [TK]D-Fender | MadMoney: Sorry, your plan has more holes than a #9 sponge |
| 04:18.24 | MadMoney | Hacked up programming. |
| 04:18.42 | MadMoney | SDL sound and C programming language. :D |
| 04:19.05 | WindowsUser | MadMoney: just break down and say you're doing it for shits and giggles |
| 04:19.09 | MadMoney | But anyway, I'm going to just buy an ATA. |
| 04:19.23 | carrar | I don't believe you |
| 04:19.26 | [TK]D-Fender | MadMoney: And what about the cost of the computer used around this? I suppose that isn't factored in. What about the cost of POWERING it? |
| 04:20.03 | WindowsUser | MadMoney: and what about the wasted electrons? seriously! save the electrons |
| 04:20.20 | MadMoney | STOP IT! |
| 04:20.27 | [TK]D-Fender | hosts a sit-down and hands out t-shirts "SAVE THE ELECTRONS!" |
| 04:20.29 | MadMoney | I said I've abandoned the idea. |
| 04:20.43 | MadMoney | will buy ATAs |
| 04:20.48 | [TK]D-Fender | MadMoney: Yes, but the entertainment value lives on in perpetuity! |
| 04:21.07 | MadMoney | http://www.youtube.com/watch?v=sb8kI3BdOog - This is more entertaining |
| 04:21.16 | [TK]D-Fender | MadMoney: You simply can't make this kind of shit up... it just "happens"! |
| 04:21.44 | MadMoney | Good thing you don't know I am Gremlin... I would never live that down. |
| 04:22.17 | [TK]D-Fender | Gremlin: You are so smart... SMRT |
| 04:23.26 | [TK]D-Fender | Gremlin: Now aside from your hapless MacGuyver complex, did you actually have something you needed to acheive in there? |
| 04:23.44 | WindowsUser | hey dont knock macguyver |
| 04:23.59 | WindowsUser | you'll have stargate fans and retired people after you |
| 04:24.03 | [TK]D-Fender | WindowsUser: I' not. I'm knocking his half-ass attempt at it |
| 04:24.28 | WindowsUser | he didn't actually build anything yet |
| 04:24.31 | WindowsUser | so...... |
| 04:24.35 | [TK]D-Fender | WindowsUser: And thats what his master plan was missing : A Swiss-Army knif, some chewing gum, aluminum foil, and a paper-clip |
| 04:25.31 | [TK]D-Fender | MadMoney: This ought to do : http://tinyurl.com/496svm |
| 04:25.40 | carrar | TK you going to astericon this year? |
| 04:26.07 | [TK]D-Fender | carrar: Never been. Too far & too costly |
| 04:26.15 | carrar | ah |
| 04:26.43 | carrar | Digium should payfor a ticket for you |
| 04:26.54 | [TK]D-Fender | carrar: File will be in town next week, so I'm going to see about getting a bunch of us together |
| 04:27.16 | [TK]D-Fender | carrar: LOL.... that.. would not work so well |
| 04:27.50 | carrar | complicity is wonderful thing |
| 04:27.53 | carrar | heh |
| 04:28.15 | [TK]D-Fender | carrar: So is "conflict of interest" ;) |
| 04:28.16 | MadMoney | goes back to fighting with Dell to get a new AC adapter. |
| 04:28.37 | [TK]D-Fender | MadMoney: Just couple 2 other ones together ;) |
| 04:28.51 | MadMoney | :@ |
| 04:33.33 | carrar | MadMoney, just tell Dell your old one caught on fire |
| 04:35.19 | MadMoney | My old one is covered by the recall but they are saying no. |
| 04:36.00 | WindowsUser | if its covered how can they say no? they send you the newer one for free? |
| 04:36.48 | MadMoney | They said the recall doesn't cover my Latitude. |
| 04:37.03 | carrar | call the cops |
| 04:37.38 | MadMoney | Cops aren't going to do anything here. The Attorney General might help. |
| 04:37.54 | MadMoney | I'm just going to buy another adapter. |
| 04:38.39 | [TK]D-Fender | MadMoney: the correct answer is "Listen I'm a happy customer. You want me to remain one. Because if I'm not, the bad press I will cause out of pure spite will cost you FAR more that the pittance of a replacement for my DEFECTIVE adapter." |
| 04:45.03 | *** join/#asterisk skanker74 (n=skanker7@124.104.186.157) |
| 04:49.33 | *** join/#asterisk skanker74 (n=skanker7@124.104.186.157) |
| 04:50.53 | *** part/#asterisk skanker74 (n=skanker7@124.104.186.157) |
| 04:55.25 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 04:56.05 | Nugget | It's tricky to rock a rhyme that's right on time. |
| 04:56.13 | *** join/#asterisk metfan2007 (n=metfan20@189.180.218.146) |
| 04:59.30 | metfan2007 | Hi all!! I have a question, I have asterisk box "A" with T1 links, and asterisk box "B", with a predictive dialer, the box "B" sends the calls to "A" via IAX2 trunk, and "A" connect them to PSTN, the question is if do I have to indicate something special in order to box "B" handle all the call PSTN status, I mean, witch answered status does box B takes? the A response or the PSTN response? |
| 05:00.59 | [TK]D-Fender | metfan2007: Depends what A tells B |
| 05:03.13 | metfan2007 | [TK]D-Fender: like what? is there any way to pass the A statues to B? I'm a little confused |
| 05:04.27 | [TK]D-Fender | metfan2007: Status is passed if you do't do something stupid like ANSWER it at A |
| 05:05.31 | ManxPower | Asterisk's HANGUPCAUSE is more or less based on Q.931 PRI hangup causes |
| 05:05.49 | metfan2007 | [TK]D-Fender: ok ok, in "A" I only have an exten => XXXX,1,Dial(${OUTSIDE}/${EXTEN} bla bla bla, is that enough? |
| 05:06.39 | [TK]D-Fender | metfan2007: Go try. |
| 05:07.02 | [TK]D-Fender | bed time, checking out. later all |
| 05:11.29 | *** join/#asterisk jpsharp (n=jsharp@24.224.45.160) |
| 05:12.18 | jpsharp | Can Asterisk send out subscribe messages to query remote servers on MWI/Voicemail? |
| 05:16.02 | MadMoney | Why would you want to do that? |
| 05:16.17 | *** join/#asterisk errotan (n=errotan@5403E519.catv.pool.telekom.hu) |
| 05:17.56 | jpsharp | Because I have Asterisk set up at home configured on one of my voip providers that also provides voicemail services. If a call comes in and my asterisk box is unreachable for some reason, the call goes to their voicemail system. |
| 05:18.12 | jpsharp | So I'd like to know if I get a voicemail there. |
| 05:18.30 | jpsharp | without having to say "Hey, I should check my voicemail box to see if I missed anything" |
| 05:25.42 | *** join/#asterisk shido6 (n=shido6@74-132-202-71.dhcp.insightbb.com) |
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| 05:59.08 | *** join/#asterisk Chris-NB (n=chris@85-126-61-10.work.xdsl-line.inode.at) |
| 06:00.08 | Zuchmir2 | following instructions from http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf (section [mysipprovider-out] ) , yet i get " chan_sip.c: Received response: "Forbidden" from ..." (I can't do registration) |
| 06:01.04 | carrar | ~book |
| 06:01.05 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
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| 06:02.44 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.215) |
| 06:17.41 | Zuchmir2 | followed instructions on page 98, but stil 403 forbidden :-( |
| 06:20.33 | *** join/#asterisk skanker74 (n=skanker7@124.104.186.157) |
| 06:21.30 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 06:22.27 | Zuchmir2 | http://pastebin.com/m5cb5da21 |
| 06:23.14 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 06:24.13 | Zuchmir2 | http://pastebin.com/d6935fac8 |
| 06:24.50 | Zuchmir2 | i have another vsp that that conf works, but this one gives 403 |
| 06:25.12 | *** join/#asterisk xrmx__ (n=rm@host129-254-dynamic.2-87-r.retail.telecomitalia.it) |
| 06:26.39 | AlmightyOatmeal | Zuchmir2: did you sacrifice the prerequisite number of kittens prior to reloading asterisk? |
| 06:30.28 | Zuchmir2 | test system, i killed *, and restarteed |
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| 06:50.24 | tzafrir_laptop | Zuchmir2, you sacrifised Asterisk ??? |
| 06:52.13 | Zuchmir2 | yeah, i'm trying to get it to work |
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| 07:34.41 | *** join/#asterisk k4mi1 (n=qmpelkam@xdsl-1814.wroclaw.dialog.net.pl) |
| 07:34.47 | k4mi1 | hi all :) |
| 07:35.57 | k4mi1 | do You know if it is possible to skip OGM in voicemail? |
| 07:37.28 | k4mi1 | I have a phone where I can set username and password so asterisk should take those credentials instead ask me to dial from phone keypad |
| 07:37.31 | k4mi1 | is it possible? |
| 07:44.18 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
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| 07:55.58 | ISO9001 | er.. I have no idea what your phone does with the username/password |
| 07:56.12 | ISO9001 | you can have asterisk drop you right into a mailbox with no authentication, but that's almost certainly not a good idea. |
| 07:56.24 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
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| 08:09.11 | k4mi1 | hmm... so it is not possible to skip this nice woman's voice "mailbox... password..."? :) |
| 08:10.01 | joobie | yes |
| 08:10.07 | joobie | check the flags |
| 08:17.19 | *** join/#asterisk luckyaba (n=Lucky@ip72-194-215-55.sb.sd.cox.net) |
| 08:19.23 | ISO9001 | k4mi1: read what I said again. You can do it, but it's very likely not a good idea. |
| 08:21.57 | wdoekes | that really depends on his usage |
| 08:22.13 | wdoekes | if there is no access from the outside.. |
| 08:22.37 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 08:26.49 | k4mi1 | I just want that asterisk take my username and password from my phone instead asking me about this |
| 08:27.41 | henk | it can't. |
| 08:27.55 | k4mi1 | aha |
| 08:28.22 | k4mi1 | do You know what kind of sip server can do this? |
| 08:28.32 | henk | unless the phone uses a clever mechanism so asterisk doesn't have to know about, which i very much doubt. what kind of phone are you talking about anyway? |
| 08:29.19 | henk | that'd be a good question for the vendor of the phone. that feature should be good for something or it wouldn't be there. perhaps ;) |
| 08:29.35 | joobie | err |
| 08:29.37 | joobie | it can |
| 08:29.47 | joobie | read the freaken manual k4mi1 |
| 08:30.01 | joobie | why do you ask a Q and not take advice? |
| 08:30.10 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 08:30.10 | joobie | VoiceMailMain([[s]mailbox]@context) |
| 08:30.17 | joobie | let's you specif the mailbox |
| 08:30.28 | joobie | the 's' flag supresses the passcode |
| 08:30.34 | joobie | tard. |
| 08:30.56 | henk | joobie: did you actually read the question? he wants to use the values he configured in the phone! |
| 08:31.20 | joobie | so? configure it with the dialplan |
| 08:31.25 | joobie | that's taking it from the phone.. |
| 08:31.31 | henk | how? |
| 08:31.35 | joobie | omg |
| 08:31.46 | joobie | it's tard hour in #asterisk |
| 08:32.12 | henk | doesn't know of a dialplan application asking a phone for some value which gets answered _by_ _the_ _phone_ and not by the user operating the phone and punching digits. enlighten us! |
| 08:32.17 | henk | :) |
| 08:33.12 | henk | k4mi1: will i get an answer to my question? |
| 08:33.28 | k4mi1 | about vendor? |
| 08:33.36 | henk | yes |
| 08:33.38 | k4mi1 | this is a new phone that is not in the market |
| 08:33.57 | joobie | henk, the phone can trigger a specific point in the dialplan.. do you know how to do this? |
| 08:34.10 | k4mi1 | I am tester of this prototype |
| 08:34.10 | joobie | if you do, the rest is self explainatory with the above.. |
| 08:34.58 | joobie | you don't need to have the phone hand a password to the dialplan .. you can just supress the password and specify the mailbox |
| 08:35.03 | joobie | problem solved. |
| 08:36.13 | k4mi1 | so password will not be needed? |
| 08:36.27 | joobie | no you tard, for the 3rd time, read the manual.. you can supress it |
| 08:36.38 | k4mi1 | ok I will read it |
| 08:36.44 | joobie | congrats. |
| 08:36.49 | k4mi1 | ;) |
| 08:37.09 | henk | joobie: don't call anyone a tard if you answer a question that was never asked. |
| 08:37.23 | joobie | <PROTECTED> |
| 08:37.23 | joobie | <PROTECTED> |
| 08:37.23 | joobie | <PROTECTED> |
| 08:37.33 | joobie | what don't you understand about that henk? |
| 08:37.40 | joobie | don't talk shit and say asteirsk cant do things that it can.. |
| 08:37.43 | henk | 0937 < k4mi1> I have a phone where I can set username and password so asterisk should take those credentials instead ask me to dial from phone keypad |
| 08:38.08 | henk | that was the original question. he was asking specifically about that feature. you need to look at the context. |
| 08:38.25 | joobie | the original question si what i pasted |
| 08:38.27 | joobie | read the backlog |
| 08:39.01 | henk | joobie: YOU read the backlog... the question you quoted was asked 20 minutes after the one i quoted. |
| 08:39.18 | henk | EOD, i have to work. no time for tards like you :-p |
| 08:39.23 | k4mi1 | hey guys do not argue :) |
| 08:39.37 | joobie | and if you read the first question as well as that proceeding comment he made, you would understand that he's just trying to get past the password prompt.. not actually submit some pass from his phone |
| 08:39.39 | joobie | you retard. |
| 08:40.13 | henk | joobie: he is testing a new phone. he wants to test that feature. he is asking for a sip server able to use that feature. |
| 08:40.26 | joobie | henk, something is wrong with your irc client, what you pasted was well after .. |
| 08:40.27 | joobie | dood |
| 08:40.29 | joobie | lick my nutsack |
| 08:40.32 | joobie | you tard |
| 08:40.39 | henk | *plonk* |
| 08:40.44 | joobie | and dont give false advice |
| 08:40.49 | joobie | asterisk cant do it he says.. pfft |
| 08:41.05 | joobie | even IF the above was not in context, im sure it could.. you just talk shit, so refrain from talking in future.. |
| 08:41.07 | joobie | tard |
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| 08:51.39 | *** mode/#asterisk [+o jtodd] by ChanServ |
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| 09:11.33 | *** join/#asterisk Balistic (n=irssi@dsl-242-208-110.telkomadsl.co.za) |
| 09:12.48 | Balistic | Hi, if 'sip show peers' show UNREACHABLE for all phones, does that only mean that there is a network/communication issue? This is happening on a LAN with only 1 switch and the server can ping all the phones. |
| 09:16.33 | *** join/#asterisk riyadc (n=riyadc@test20.citech-bd.com) |
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| 09:41.49 | angryuser | Balistic, try to set qualify=no and call |
| 09:47.27 | Balistic | angryuser: wont that create other problems? |
| 09:47.59 | angryuser | For example ? |
| 09:48.42 | angryuser | Balistic, you cant call right ? do you think it can be worse ? |
| 09:49.13 | Balistic | im not interested in quick fixes, id rather solve the actualy problem |
| 09:49.54 | Balistic | -y |
| 09:52.16 | angryuser | fix then |
| 09:53.41 | Balistic | But I am struggling to actually diagnose the problem, that is why I am asking for assistance. |
| 09:58.46 | NoxIn- | Balistic: the phones are registering with asterisk as a proxy, or the are all trunk with fixed IP address ? |
| 09:59.53 | Balistic | Each phone receives a reserved IP via dhcp, and registers itself to the asterisk server using a extension+secret. |
| 10:00.31 | Balistic | These are Snom320 phones |
| 10:00.31 | NoxIn- | and do you see the trace of the registration ? |
| 10:01.05 | Balistic | I have not looked for the registration specifically, but I did see SIP traffic in both directons. |
| 10:01.41 | Balistic | via a ngrep on the server. I am not at the client currently, but will check for that when I am there again later. |
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| 10:11.50 | NoxIn- | unreable means either network problem (but if you see sip traffic in both direction it should be ok) or the phone don't respond to qualify resuqest |
| 10:12.13 | NoxIn- | if you dump sip traffic the qualify are "SIP OPTION requests" |
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| 12:05.56 | *** join/#asterisk dominic1 (n=Miranda@213.221.82.242) |
| 12:09.01 | dominic1 | Short question. I want to build a mediagateway. Is there any possibility to build a failover system with multiple servers on one ISDN - NTBA? |
| 12:09.45 | *** part/#asterisk LemensTS (n=customgt@70.238.180.76) |
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| 12:12.15 | dominic1 | I am thinking about using a 1.4 version of asterisk or a 1.6 version for my mediagateway |
| 12:15.12 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
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| 12:31.22 | ariel_ | hello everone |
| 12:31.50 | *** join/#asterisk LtScarr (i=benno@palm.hoeg.nl) |
| 12:32.04 | LtScarr | hey everyone |
| 12:32.16 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 12:33.19 | ariel_ | anyone here used the new Xorcom Live CD 1.3 ? |
| 12:33.37 | LtScarr | can somebody tell me if the cheap w6692cf is qualified for handling calls through asterisk? |
| 12:34.51 | tzafrir_laptop | votes :-) |
| 12:35.13 | LtScarr | aka Dynalink IS64PH |
| 12:35.25 | ariel_ | LtScarr: I have no idea on what it is. |
| 12:35.47 | dominic1 | I am using Asterisk 1.6 in my environment and want to build up a mediagateway. I am using G722 internally. Should i now use Asterisk 1.4 on the gateway or 1.6? Can i get trouble with the codecs in 1.4, cause I am getting G711 from the ISDN network? Or will my primary Asteriskserver fall back to G711 if it's in his codecorder? |
| 12:35.49 | LtScarr | it's the first time i'm working with ISDN |
| 12:36.02 | ariel_ | tzafrir_laptop: your name is in allot of the files for the live cd. Can it or does it have an option to install on a hdd to run some of your asteriskbanks? |
| 12:36.33 | LtScarr | it is a ISDN modem card |
| 12:36.33 | tzafrir_laptop | ariel_, not really. but you can just use a Debian Lenny system with those files |
| 12:37.23 | ariel_ | OK I have most of our systems on Debian Lenny, any easy way to move them? |
| 12:37.27 | *** join/#asterisk Nasra (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
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| 12:41.10 | [TK]D-Fender | dominic1: If you are using a second server for termination then choose where you want the transcoding to take place. Either on your phone's server, or the termination one. |
| 12:42.54 | [TK]D-Fender | LtScarr: Get Googling. I've never head of that brand before |
| 12:42.58 | [TK]D-Fender | heard* |
| 12:43.03 | dominic1 | @[TK]D-Fender Thank you for your answer! |
| 12:45.11 | tzafrir_laptop | ariel_, check the apt source in /etc/apt/sources.list |
| 12:45.51 | ariel_ | tzafrir_laptop: great t/y |
| 12:46.12 | tzafrir_laptop | I generally also need to publish the full config for the live CD, which should be done when the final version is to be released |
| 12:46.21 | tzafrir_laptop | (it's already part of the upload script) |
| 12:46.26 | ariel_ | just in case you know this or not, it's running the asterisk-gui 2.0.5 but has a few links that still go to freepbx |
| 12:46.46 | tzafrir_laptop | yeah, that should already be fixed |
| 12:46.59 | ariel_ | great it's a good cd by the way. |
| 12:47.17 | tzafrir_laptop | in fact I figure I should upload a "second beta" |
| 12:47.46 | ariel_ | please do, and when? I could use it for the trip I am doing. Going to take an asteriskbank to do some onboard testing. |
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| 13:18.00 | dominic1 | another question:I also receive my facsimile with asterisk. Currently I am using asterisk 1.6. In the future i want just forward the ISDN-Stuff to my asterisk via IAX and receive the facsimile on the asterisk with app_fax or hylafax. My gateway will run on asterisk 1.4, cause i don |
| 13:18.13 | dominic1 | 't need the additional features on the gateway |
| 13:19.03 | dominic1 | Any tips, if i should use asterisk 1.6 on the gate too? Should i receive the faxes on the gateway itself? |
| 13:19.17 | [TK]D-Fender | dominic1: You will be ading latency & PL risk even across a LAN to your other server which may kill a fax |
| 13:19.27 | *** join/#asterisk moa_ (n=moa_@lab.vision.net) |
| 13:20.02 | dominic1 | but my problem is, that my database including the emailaddresses is hosted on the main asterisk system |
| 13:21.17 | *** join/#asterisk ShurV (n=ShurV@62-90-161-33.barak.net.il) |
| 13:21.21 | ShurV | Hello! |
| 13:22.18 | ShurV | Please, how to hide the caller ID on asterisk as SIP client, (I found this in documentation: Asterisk sip restrictcid : (yes/no) To have the callerid restricted -> sent as ANI; use this to hide the caller ID. This does not seem to work. This variable has been deprecated as of v1.2.x.) |
| 13:24.21 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 13:24.43 | voipheroes | use the Privacy header with SipAddHeader |
| 13:24.59 | voipheroes | or set the variable CALLERID(num), CALLERID(name) |
| 13:25.10 | voipheroes | it depend of your outbound trunk |
| 13:25.21 | [TK]D-Fender | ShurV: "core show application setcallerpres" |
| 13:25.31 | voipheroes | some handle the Privacy header while some others just need to have a special Callerid number |
| 13:25.37 | voipheroes | (like ten 0) |
| 13:25.43 | voipheroes | or ("anonymous") |
| 13:25.59 | *** join/#asterisk moa_ (n=moa_@lab.vision.net) |
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| 13:31.12 | *** join/#asterisk jmkgreen (n=chatzill@84.12.140.56) |
| 13:32.14 | jmkgreen | Hi is the 1.6.1.x release train considered stable? I'm a little confused about which versions are stable and which are beta. |
| 13:32.26 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:32.53 | Katty | :> |
| 13:32.54 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 13:32.59 | Katty | hugs jaytee |
| 13:33.25 | jaytee | mornin Katty |
| 13:33.29 | Katty | herroes. |
| 13:33.31 | Katty | how're you dear |
| 13:35.48 | [TK]D-Fender | jmkgreen: 1.6.0 & 1.6.1 branches are out of beat and in full release. 1.6.0 is on release 10 and considered pretty stable. 1.6.1.1 jsut came out and I'd wait on another release or 2 before going production with it |
| 13:35.54 | [TK]D-Fender | Katty: Mew. |
| 13:36.01 | Katty | hi fender. |
| 13:38.47 | *** join/#asterisk blebleble (n=godie@s7.n194.n222.n216.static.myhostcenter.com) |
| 13:39.12 | jmkgreen | [TK]D-Fender: Thx |
| 13:42.23 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 13:42.23 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 13:43.27 | blebleble | I'm currently having a problem maybe someone can shed some light onto, when a user from our queue transfers a call out to a person (manager), that person is not getting another call from the queue until the call he transfered is completed |
| 13:46.36 | [TK]D-Fender | blebleble: This has been the case since forever. The solution is to use * DTMF transfers, and not native SIP |
| 13:48.04 | *** join/#asterisk ehsjoar (n=ehsjoar@24.9.91.203) |
| 13:48.05 | *** part/#asterisk ehsjoar (n=ehsjoar@24.9.91.203) |
| 13:48.11 | *** join/#asterisk ehsjoar (n=ehsjoar@24.9.91.203) |
| 13:48.16 | ShurV | voipheroes: how to do it within sip.conf ? (hide caller ID... I use asterisk as PSTN to SIP gw, but X-lite shows PSTN_number@astersk_IP, it's not good) |
| 13:48.29 | blebleble | [TK]D-Fender: is this something easy to complete? |
| 13:48.35 | *** join/#asterisk tobias (n=tobias@71.70.219.40) |
| 13:49.32 | [TK]D-Fender | ShurV: What X-Lite shows is irrelevant. That is a separate leg from the one from * to your gateway |
| 13:49.43 | [TK]D-Fender | blebleble: "core show application queue" <- |
| 13:50.13 | ShurV | [TK]D-Fender: it very relevant, this is the problem |
| 13:51.13 | [TK]D-Fender | ShurV: the leg from X-Lite to * is completely separare. It hits the dialplan and you can do whatever you want before calling out your gateway. * si not a proxy, it is a B2BUA. |
| 13:51.20 | ShurV | if I call my server from the phone, it forwards me to sip client (xlite for instance), and the sip client shows the ID |
| 13:51.26 | [TK]D-Fender | ShurV: So don't pin this on "SIP caller ID". |
| 13:51.47 | [TK]D-Fender | ShurV: Your dialplan can manipulate things however you want <- |
| 13:51.49 | ShurV | I didn't |
| 13:52.26 | [TK]D-Fender | ShurV: call comes in looking like X, goes out looking like Y. |
| 13:52.34 | ShurV | I told to hide PSTN caller ID from SIP nertwork |
| 13:52.47 | [TK]D-Fender | ShurV: Told what? |
| 13:53.25 | voipheroes | ShurV, : it's not within sip.conf but rather inside your dialplan |
| 13:53.28 | ShurV | asterisk forwards calls coming through E1 from PSTN to SIP |
| 13:54.14 | voipheroes | [TK]D-Fender is right |
| 13:54.24 | [TK]D-Fender | ShurV: * make process a call from E1 and choose to dial a SIP device, but what it presents you can manipulate |
| 13:54.48 | ShurV | Actually that what I ask |
| 13:55.50 | voipheroes | http://www.voip-info.org/wiki/view/Asterisk+func+callerid |
| 13:56.10 | ShurV | how to manipulate the forwarding of (pstn) caller id and the (gw) ip to be hidden from callee sip client? |
| 13:59.00 | ShurV | I'm afraid I cannot use the extensions.conf |
| 13:59.06 | [TK]D-Fender | [09:25]<[TK]D-Fender>ShurV: "core show application setcallerpres" |
| 13:59.10 | [TK]D-Fender | ShurV: And why not? |
| 13:59.24 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 14:03.17 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:03.17 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:03.34 | *** part/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
| 14:05.57 | Katty | anyone having an issue with the us.pool.ntp.org servers? |
| 14:06.10 | ShurV | cause I'm quite noob, and I don't understand even how to use extensions if a caller on this line doesn't do dial any extension |
| 14:06.33 | [TK]D-Fender | ShurV: then how is your SIP device even getting called? |
| 14:06.51 | *** join/#asterisk flashnet (n=fdfdsf@201-213-121-79.net.prima.net.ar) |
| 14:06.54 | ShurV | cause I'm not familiar with astereisk at all, I just build applications in php for the IVR using an API |
| 14:06.57 | [TK]D-Fender | ShurV: You shove a line before the dial and that it. this isn't Raw-Cat Sigh-Hence |
| 14:07.21 | jamesh1 | Question: users logged into a queue not getting logged out after missing a call. -- autologoff=14 , isn't that the only thing needed? |
| 14:07.43 | *** join/#asterisk javierj (n=kvirc@190.146.192.22) |
| 14:08.03 | ShurV | I have a funcion in API - SP_DIal([trunkid], [number or SIP addres]...) |
| 14:08.05 | [TK]D-Fender | jamesh1: Last I checked only those using AgentLogin can be kicked like that... |
| 14:08.31 | javierj | hi everyone |
| 14:08.31 | [TK]D-Fender | ShurV: When a call comes in it his the dialplan, not some other random API |
| 14:09.29 | javierj | How can I reduce to the minimum the time an incoming call get to the dialplan context?? |
| 14:09.53 | jamesh1 | [TK]D-Fender: im just using 'agent-add' apparently. |
| 14:10.01 | ShurV | [TK]D-Fender: yes, there are some settings predefined in extensions.conf by the API builder. I just afraid to mess with them |
| 14:10.02 | jamesh1 | example: == Spawn extension (macro-agent-add, s, 3) exited non-zero on 'SIP/8149-c803d5a0' in macro 'agent-add' |
| 14:11.06 | [TK]D-Fender | jamesh1: that is some random bit of dialplan that doesn't tell us what you're doing. And from the loko of things, I'm getting the impression you don't either. |
| 14:11.32 | angryuser | good day, can someone suggest me a free softphone with a "Transfer" button ? thanks |
| 14:11.33 | ShurV | Well, thankyou, I'll try to contact the API developer, there must be some upper level function |
| 14:11.34 | [TK]D-Fender | ShurV: Well if you're not in control of your own system then there isn't much to say at this point. |
| 14:11.51 | [TK]D-Fender | angryuser: Ekiga / Zoiper |
| 14:11.56 | ShurV | I understand |
| 14:12.19 | [TK]D-Fender | javierj: Huh? |
| 14:12.20 | jamesh1 | d-fender: agent-add is just a phone logging into a queue(extension) |
| 14:12.25 | ShurV | <[TK]D-Fender: I just thought that there |
| 14:12.31 | jamesh1 | and you're right, I have no clue what I am doing. |
| 14:12.36 | ShurV | might b config option |
| 14:12.40 | [TK]D-Fender | jamesh1: that is jsut some macro name. It does not show what it is actually doing |
| 14:12.51 | jamesh1 | I know what it does. |
| 14:13.00 | jamesh1 | it adds the extension to the member list of the queue |
| 14:13.05 | [TK]D-Fender | ShurV: Your system is controlled by some GUI. IT is in charge and IT has to offer you "options". |
| 14:13.45 | ShurV | True |
| 14:13.53 | javierj | [TK]D-Fender: Thanks, what I mean is I'd like to reduce the time asterisk wait when it detects an incoming call in a zap channel befor it passes the control to dialplan |
| 14:14.39 | [TK]D-Fender | javierj: Most of the time * waits for CALLERID <- before starting to process the call. disable that if you want and * won't wait for it |
| 14:15.20 | [TK]D-Fender | jamesh1: Well that isn't AgentLogin so AFAIK that kick option won't work. they are static memebers. |
| 14:15.35 | javierj | [TK]D-Fender: thanks.. I'll do it.. I need to proccess calls as soon as posible... thanks |
| 14:15.52 | [TK]D-Fender | jamesh1: If you want to kick them you'll have to add your own code in dialplan calling to a local channel |
| 14:16.14 | [TK]D-Fender | javierj: What I jsut mentioned of course only applies to analog, not digital |
| 14:16.19 | jamesh1 | [TK]D-Fender: Yea I was trying to get around setting up all the agentcallbacklogin's |
| 14:16.49 | [TK]D-Fender | jamesh1: You can't log out a device. "Agents" log in, not devices |
| 14:17.13 | javierj | [TK]D-Fender: Ok |
| 14:18.00 | jamesh1 | [TK]D-Fender: I understand what you are saying, but if a user calls the login extension and dials the wrong extension. the queue has a wrong number in the list and then it gets 'calls' which obviously turn up invalid. |
| 14:18.20 | jamesh1 | and adds time to the overall hold time |
| 14:18.42 | [TK]D-Fender | jamesh1: Well I guess you'd better VALIDATE your input :) |
| 14:18.43 | jamesh1 | I was just thinking there has to be a way to define a timeout for that. |
| 14:18.50 | jamesh1 | lol yea |
| 14:19.20 | jamesh1 | [TK]D-Fender: I found out that 'bug' the hard way |
| 14:19.25 | [TK]D-Fender | jamesh1: Part of having such great control over your system is the ability to tell it to do something stupid. Don't complain when it lets you :) |
| 14:19.35 | jamesh1 | bug in my setup. |
| 14:19.41 | jamesh1 | exactly |
| 14:20.13 | angryuser | [TK]D-Fender, zoiper ask money for evey button :( |
| 14:20.38 | jamesh1 | I got thrown a project and it turned out to be the phone system, so I'm learning asterisk from nothing. :( |
| 14:21.04 | [TK]D-Fender | jamesh1: \o/ |
| 14:21.09 | *** join/#asterisk dni (n=dniz0r@adsl-074-169-015-252.sip.mia.bellsouth.net) |
| 14:21.32 | jamesh1 | well I have the new system up, just was never able to fully load test so working out all the kinks now. |
| 14:21.44 | jamesh1 | pain in the butt |
| 14:21.59 | *** join/#asterisk jeffik (n=jeffik@69-196-165-181.dsl.teksavvy.com) |
| 14:22.38 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 14:23.02 | jamesh1 | I setup Elastix and regret it. |
| 14:23.45 | [TK]D-Fender | jamesh1: I'm sorry... it's too lat for you now... |
| 14:23.56 | [TK]D-Fender | jamesh1: Soul has been sold to the lowest bidder.... |
| 14:23.59 | [TK]D-Fender | late* |
| 14:24.01 | jamesh1 | lol |
| 14:24.24 | dni | Hello Everyone,. If someone has a moment could they please help me with a one way audio issue im having,.. Im trying to provide a feature to our other office that we have via CCM-Asterisk SIP trunk,. . when they dial out the asterisk box to a cell, there is one way audio,. So there is a routing issue somewhere but im not sure where ... here is a pastebin of the global debug http://pastebin.com/m32635e1c |
| 14:24.47 | jamesh1 | well our previous system was just asterisk 1.2 on some very old hardware and calls were getting noise/static |
| 14:25.39 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 14:26.27 | [TK]D-Fender | dni: make sure to disable reinvites globally on your * |
| 14:26.39 | *** join/#asterisk Joel (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 14:26.40 | [TK]D-Fender | dni: Enpoints may attempt to reconnect and not have a route |
| 14:28.21 | *** join/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
| 14:28.31 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 14:28.50 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 14:30.15 | jamesh1 | [TK]D-Fender: is there a preferred interface? |
| 14:30.26 | [TK]D-Fender | jamesh1: vi <- |
| 14:30.33 | jamesh1 | lol I use vi |
| 14:30.52 | jamesh1 | we were tired of editing 10 files to add a new user |
| 14:31.35 | [TK]D-Fender | jamesh1: Usually 3-4 tops |
| 14:31.50 | [TK]D-Fender | jamesh1: Oh well. |
| 14:32.04 | jamesh1 | Well I'm definitely regretting it now. |
| 14:32.11 | [TK]D-Fender | jamesh1: When you hire a chauffeur, don't complain how he drives |
| 14:32.19 | jamesh1 | Might make a parallel build right now lol |
| 14:32.22 | dni | [TK]D-Fender, thanks |
| 14:32.27 | dni | seemed to work |
| 14:32.34 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733615.dsl.bell.ca) |
| 14:32.35 | [TK]D-Fender | dni: You're welcome |
| 14:37.48 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:38.50 | *** join/#asterisk errotan (n=errotan@5403E753.catv.pool.telekom.hu) |
| 14:45.01 | *** join/#asterisk blkry (n=blkry@64.147.222.130) |
| 14:50.01 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:50.47 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 14:52.18 | tzafrir_laptop | ariel_, http://updates.xorcom.com/iso/live-2.0.0-beta.img (usb disk image) and http://updates.xorcom.com/iso/live-2.0.0-beta.iso |
| 14:52.47 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 14:53.31 | tzafrir_laptop | The DebianLive config I used is under http://updates.xorcom.com/iso/live-2.0.0-beta_config/ |
| 14:53.58 | verywiseman | where can i know registration cost to sip proxy? |
| 14:54.46 | [TK]D-Fender | verywiseman: huh? |
| 14:55.10 | verywiseman | [TK]D-Fender, what is problem? |
| 14:55.30 | [TK]D-Fender | verywiseman: Problem is figuring out exactly what you're asking... |
| 14:57.57 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 14:58.15 | *** join/#asterisk Maxxed (n=max@216.215.95.114) |
| 15:01.07 | *** join/#asterisk Joel (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 15:02.14 | asaleem | asterisk boxe's sip is bound to public IP but the actual phones are behind NAT. Do I need nat=yes in sip.conf? |
| 15:03.49 | [TK]D-Fender | asaleem: which NAT? |
| 15:04.46 | asaleem | [TK]D-Fender, PAT |
| 15:05.36 | [TK]D-Fender | asaleem: No, I mean are the phones local to *? |
| 15:05.40 | asaleem | [TK]D-Fender, Simple one, just to share the inetrnet |
| 15:06.06 | asaleem | [TK]D-Fender, no the phones are not local |
| 15:06.25 | asaleem | Phones - NAT server - * |
| 15:06.30 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 15:07.05 | [TK]D-Fender | asaleem: Then yes, * should have "nat=yes", "qualify=yes for every remote phone |
| 15:08.00 | asaleem | [TK]D-Fender, if I put them in [general], that would suffice, I guess? |
| 15:08.10 | beek | mornin' [TK]D-Fender |
| 15:08.36 | jaytee | mornin' beek |
| 15:08.42 | beek | morning jaytee |
| 15:08.54 | [TK]D-Fender | asaleem: DON'T. General settings are just that... this can screw up ITSP entries that should NOT be "nat=yes" |
| 15:09.07 | [TK]D-Fender | beek: *yawn* |
| 15:11.16 | *** join/#asterisk Von_Lorenz (n=lorenzo@ip-89-162.sn1.eutelia.it) |
| 15:11.21 | Maxxed | man asterisk has changed alot since i last fooled with it ;) |
| 15:11.29 | asaleem | [TK]D-Fender, I changed them for each phones. Now it happens that if phones 401 to 406 work, then 407 & 408 don't work and vice versa |
| 15:11.34 | Von_Lorenz | Hi guys |
| 15:11.37 | beek | Maxxed: For the better... |
| 15:11.41 | asaleem | [TK]D-Fender, any thoughts |
| 15:11.42 | Maxxed | looks like it! |
| 15:11.53 | Maxxed | i gota brush up on the ol foo |
| 15:12.08 | [TK]D-Fender | asaleem: Are these phones all behind the SAME NAT? |
| 15:12.16 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 15:12.17 | asaleem | [TK]D-Fender, yes |
| 15:12.30 | [TK]D-Fender | asaleem: Change their SIP port so taht they are unique |
| 15:13.01 | asaleem | [TK]D-Fender, on the phones? |
| 15:13.03 | bmoraca | asaleem: sounds like your NAT router isn't handling SIP properly (or isn't powerful enough for the number of phones you have) |
| 15:13.10 | [TK]D-Fender | asaleem: yes |
| 15:13.54 | asaleem | bmoraca, I did not thought abt that |
| 15:14.25 | asaleem | [TK]D-Fender, they are xlite soft phones, dont seem to find the place to change them |
| 15:15.22 | Zhad | or maybe a sip proxy |
| 15:15.41 | [TK]D-Fender | ^^^ |
| 15:15.50 | bmoraca | asaleem: a combination of "sip show peers" and enabling sip debug should let you know. If you get retransmits for the phones that stop working, then you've got a problem with your router not keeping the NATs alive long enough or recycling the ports. asterisk needs to be configured with nat=yes, but your router also needs to be configured properly |
| 15:16.05 | *** join/#asterisk ramindia (n=balajibh@202.63.96.10) |
| 15:16.32 | asaleem | bmoraca, let me check |
| 15:17.49 | Maxxed | anybody know off hand how to make the "Line 2" button on a polycom ip330 the voicemail/messages button? |
| 15:18.03 | Maxxed | something you add to the sip.cfg, but i cant seem to find it |
| 15:18.28 | Maxxed | iv never really used polycoms, always ciscos |
| 15:18.42 | Zhad | doesn't the 330 have a messages button? |
| 15:18.53 | [TK]D-Fender | Maxxed: its in the admin guide, and there is even a dedicated doc for this on their site |
| 15:19.07 | [TK]D-Fender | Maxxed: Or you can jsut use 1 line key for your reg and use a Directory contact for VM |
| 15:19.12 | Maxxed | [TK]D-Fender: thanks, il try and find it |
| 15:19.36 | Maxxed | [TK]D-Fender: yeah, thats an option, but the messages light would be nice |
| 15:21.51 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
| 15:24.25 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.97) |
| 15:25.18 | Maxxed | ah hah! add (((key.IP_330.10.function.prim = âMessagesâ))) to the <key/> mammy jammer in the sip.cfg :) |
| 15:25.28 | Maxxed | with out the ((()))'s of course |
| 15:26.11 | asaleem | bmoraca, where would look for retransmits? |
| 15:26.21 | bmoraca | sip set debug |
| 15:27.00 | [TK]D-Fender | Maxxed: Which key did you hijack for it? |
| 15:27.04 | Maxxed | Line 2 |
| 15:27.34 | Qwell | They don't have a dedicated messages button? I thought all Polycoms did O.o |
| 15:27.35 | [TK]D-Fender | Maxxed: I'd jsut as soon se the directory method. makes the function dynamic and so you don't need to change provisioning to adapt it |
| 15:27.47 | [TK]D-Fender | Qwell: IP 3XX never did |
| 15:27.53 | Qwell | oh |
| 15:28.10 | *** join/#asterisk MoreAllLess (n=Justo@cpe-76-169-252-172.socal.res.rr.com) |
| 15:28.11 | [TK]D-Fender | Qwell: I modded one of the useless buttons on my 301's for this |
| 15:28.12 | *** join/#asterisk Norman_ (n=chatzill@189.27.252.40.dynamic.adsl.gvt.net.br) |
| 15:29.09 | Maxxed | i got ya [TK]D-Fender, but the idea to have a messages light is nice |
| 15:29.12 | Norman_ | hello |
| 15:29.27 | Maxxed | using the line 2 button i would expect would give that |
| 15:29.42 | bmoraca | Maxxed: the MWI is independent of your line appearance key configuration. |
| 15:29.45 | [TK]D-Fender | Maxxed: MWI has nothing to do with setting a button. |
| 15:29.46 | Maxxed | or is there some other way to show that a voicemsg is waiting |
| 15:29.57 | [TK]D-Fender | maxx* NOTIFY's for this |
| 15:30.02 | Maxxed | oh heck, then maybe this isnt needed at all :) |
| 15:30.10 | Von_Lorenz | exit |
| 15:30.34 | Maxxed | im new to the polycom mess, and have been out of the asterisk scene for a few years |
| 15:30.52 | [TK]D-Fender | Maxxed: You seem quite bright so far. I'm sure you'll do jsut fine |
| 15:31.05 | Maxxed | eh, im good at cowboying my way thru stuff ;) |
| 15:31.36 | beek | Maxxed: If it jams, force it. If it breaks it needed replaced anyway. |
| 15:31.46 | Maxxed | hah ;) |
| 15:31.51 | Norman_ | If someone can please throw some light over my head about external SIP calls entering by ip without authentication/registering |
| 15:32.19 | bmoraca | Norman_: you have a context in your [general] section of SIP.conf that actually does something. |
| 15:32.27 | Norman_ | After searching and reading, I conclude that it was only a matter of creating |
| 15:32.28 | Norman_ | yes |
| 15:32.31 | *** join/#asterisk hfb (n=hfb@pool-98-112-240-188.lsanca.dsl-w.verizon.net) |
| 15:32.33 | Norman_ | [from-sip] |
| 15:32.35 | bmoraca | change it to a context that does nothing, and the problem will go away |
| 15:33.09 | Norman_ | but I need to catch those calls and effectively route them alternatively |
| 15:33.50 | bmoraca | then people are going to be able to send sip calls to your box without registering...that's what the context in [general] does |
| 15:33.55 | Norman_ | the problem is: even with that context at sip.conf [general] section, those calls arent falling at the [from-sip] context, but at one of the contexts defined at the sip extensions |
| 15:34.25 | bmoraca | then they're registering. change the secret to something more secret |
| 15:34.30 | Zhad | do you have context=from-sip in your [general] ? |
| 15:34.36 | Norman_ | yes |
| 15:34.44 | Norman_ | let me explain better the scenario |
| 15:34.48 | Norman_ | two asterisk boxes |
| 15:35.36 | bmoraca | two asterisk boxes should be authenticating to each other as SIP or IAX peers. you should never need the context in [general] to actually do anything...that's just bad form |
| 15:35.39 | Norman_ | on one, i want to send calls to the second one by just using Dial(559999999999@x.x.x.x,30,tT) |
| 15:35.57 | Norman_ | x.x.x.x is the ip of the second one, with the [from-sip] context at [general] |
| 15:36.40 | [TK]D-Fender | Norman_: Not smart. Auth them |
| 15:36.48 | Norman_ | hmm |
| 15:36.50 | Norman_ | right |
| 15:37.04 | bmoraca | Norman_: that's fine and good...but if you have the context in [general] actually able to route calls, people are going to be able to place calls without authenticating. you can't have it both ways |
| 15:37.05 | Norman_ | i was firewalling port 5060 between them |
| 15:37.48 | bmoraca | application-layer security superceeds layer 4 security |
| 15:38.04 | Norman_ | just a lazy man first try |
| 15:40.13 | *** join/#asterisk _bugz_ (n=bugz@adsl-99-129-29-102.dsl.lsan03.sbcglobal.net) |
| 15:41.40 | Katty | anthm: i ended up putting Rimmel London Cafe au Lait color on |
| 15:42.37 | Zhad | Norman_> If you'r eusing iax, you can use RSA authentication too. |
| 15:42.58 | Katty | anthm: http://resources.shopstyle.com/pim/46/4b/464b6a73ee34d3fbc70119cbf53751a1_medium.jpg |
| 15:44.54 | Norman_ | going to read some more and do the right way then, create an iax or sip channel at the second one and auth the first one using it to place the calls there, i guess |
| 15:45.15 | Norman_ | Thanks folks! |
| 15:45.47 | jamesh1 | With agentcallbacklogins removed in 1.6 what is the equivalent or replacement? |
| 15:49.20 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 15:49.59 | ManxPower | jamesh1: read the upgrade* files in the Asterisk source dir. |
| 15:51.35 | ManxPower | IIRC, you can create a dialplan to emulate agentcallbacklogin, that should be mentioned in the upgrade info files |
| 15:52.33 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:55.14 | *** join/#asterisk fallenstorm (n=fallen@89-178-254-42.broadband.corbina.ru) |
| 15:56.20 | jaytee | there's a file in the docs in tex format called queues-with-callback-members.tex that shows how to do agent call back without the AgentCallbackLogin app. it's AEL code but it can be adapted to standard dialplan |
| 15:56.44 | fallenstorm | hello, i have a few interfaces on my server. How can i launch asterisk on every interface? Is there some manual? |
| 15:57.47 | Qwell | fallenstorm: network interface? default configs will bind to all addresses |
| 15:59.19 | jaytee | when I've had more than one NIC enabled my callerid info on the phones got messed up and I was getting "exten"@ipaddress info instead. |
| 15:59.33 | fallenstorm | Qwell yes, network interfaces, but i need to launch the same count of asterisk applications |
| 15:59.42 | Katty | GOOD MORNING EVERYONE |
| 15:59.44 | Katty | I FINALLY WOKE UP |
| 15:59.52 | jaytee | morning Katty *hugs* |
| 15:59.54 | Qwell | fallenstorm: why? O.o |
| 15:59.56 | Katty | hugs on jaytee |
| 15:59.59 | Katty | :> |
| 16:00.16 | *** join/#asterisk kbukhari (n=kashif@119.153.65.196) |
| 16:00.23 | kbukhari | hello |
| 16:00.31 | kbukhari | i am usng astersk with ss7 |
| 16:00.40 | kbukhari | and having a littel issu |
| 16:01.07 | Katty | Qwell: YOU SIR |
| 16:01.11 | Katty | Qwell: SHALL BE HUGGED. |
| 16:01.27 | kbukhari | some time incoming call cant not play voice |
| 16:01.29 | fallenstorm | Qwell chief sayd |
| 16:01.29 | fallenstorm | =) |
| 16:01.31 | Katty | hugs on Qwell |
| 16:01.31 | Qwell | runs away screaming |
| 16:01.37 | Katty | buwaahhhahah |
| 16:01.55 | Qwell | fallenstorm: does he not understand networking or something? |
| 16:02.07 | Katty | has anyone seen mister madson lately? |
| 16:02.08 | Katty | madison? |
| 16:02.19 | Katty | leif. |
| 16:02.21 | fallenstorm | Qwell he knows networknig better than i |
| 16:02.37 | kbukhari | can i have answer about ss7 here ? |
| 16:02.48 | kbukhari | or there is any ther channel for such query ? |
| 16:03.04 | fallenstorm | Qwell is it impossible to do this? |
| 16:03.18 | Qwell | No.. it's just asking for trouble |
| 16:03.29 | fallenstorm | =) |
| 16:03.48 | Qwell | and I honestly doubt you could give a valid need to do so |
| 16:04.05 | fallenstorm | so, trust me, i need it |
| 16:04.06 | fallenstorm | =) |
| 16:04.11 | fallenstorm | -so |
| 16:04.13 | kbukhari | Qwell are u talking to me ? |
| 16:04.40 | fallenstorm | Qwell so< could you advise me some manual? |
| 16:05.08 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:05.08 | Qwell | fallenstorm: look for bindaddr in many of the sample config files |
| 16:05.52 | fallenstorm | but how can you launch one more proccess? |
| 16:05.52 | fallenstorm | asterisk -C /etc/asterisk2/asterisk.conf |
| 16:05.52 | fallenstorm | Asterisk already running on /var/run/asterisk.ctl. Use 'asterisk -r' to connect. |
| 16:05.58 | fallenstorm | it refuse me |
| 16:06.18 | Qwell | change paths in asterisk.conf |
| 16:06.26 | fallenstorm | oh |
| 16:06.32 | fallenstorm | thats a point! |
| 16:06.33 | fallenstorm | =) |
| 16:06.51 | *** join/#asterisk icyValk77 (n=icyValk7@host81-153-93-26.range81-153.btcentralplus.com) |
| 16:06.55 | fallenstorm | lamer |
| 16:06.57 | fallenstorm | =) |
| 16:07.00 | fallenstorm | tnx |
| 16:10.02 | tzafrir_laptop | fallenstorm, you also need a separate varrun directory |
| 16:10.07 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:10.12 | fallenstorm | yep |
| 16:10.15 | fallenstorm | i did it |
| 16:10.21 | fallenstorm | but nothing happens |
| 16:10.27 | tzafrir_laptop | also note that the asterisk init script assumes that there's only one asterisk instance running |
| 16:10.34 | jaytee | running multiple instances of asterisk on the same system? ugh! |
| 16:10.44 | fallenstorm | ye |
| 16:10.51 | fallenstorm | cascade-ws:~# ps xa|grep asterisk |
| 16:10.51 | fallenstorm | 18105 ? Ssl 0:00 /usr/sbin/asterisk |
| 16:10.51 | fallenstorm | 18473 pts/0 S+ 0:00 grep asterisk |
| 16:10.51 | fallenstorm | cascade-ws:~# asterisk -C /etc/asterisk2/asterisk.conf |
| 16:10.52 | fallenstorm | cascade-ws:~# ps xa|grep asterisk |
| 16:10.52 | fallenstorm | 18105 ? Ssl 0:00 /usr/sbin/asterisk |
| 16:10.54 | fallenstorm | 18486 pts/0 S+ 0:00 grep asterisk |
| 16:10.59 | *** join/#asterisk dominic1 (n=Miranda@213.221.82.242) |
| 16:11.03 | tzafrir_laptop | fallenstorm, for a quick start, see contrib/scripts/live_ast |
| 16:11.04 | jaytee | ~pastebin |
| 16:11.04 | infobot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:11.13 | fallenstorm | sor |
| 16:11.25 | tzafrir_laptop | though it will use a completely different asterisk instance |
| 16:11.29 | jaytee | we live, we learn.....or we get eaten by lions |
| 16:11.41 | fallenstorm | =) |
| 16:12.49 | *** join/#asterisk sjobeck (n=Adium@c-24-22-30-42.hsd1.or.comcast.net) |
| 16:13.50 | coppice | a few africans who didn't learn might get eaten by lions, but most of us have a less glorious end being eaten by bacteria |
| 16:14.52 | beek | jaytee: Depends on where you work if you face that threat... ;-) |
| 16:15.22 | bmoraca | yummy necrosis in the morning? |
| 16:20.58 | Maxxed | anyone know the CID variable off hand? what im looking for is when a user dials *98 VoicemailMain() there cid (4 digit sip extention) is passed |
| 16:21.06 | Maxxed | so they are just prompted for a passwd |
| 16:21.47 | Maxxed | ${CALLERID} or some such.. |
| 16:22.17 | Maxxed | is a vew versions out of date ;) |
| 16:22.40 | *** join/#asterisk Zuchmir2 (n=Zuchmir-@ool-18bd3bfc.dyn.optonline.net) |
| 16:24.18 | Zuchmir2 | i get "423 Registration Interval Too Brief" , is maxregexpire and minregexpire the ones affecting this (* is sending: "Expires: 120", response is: "Min-Expires: 3600") |
| 16:25.19 | kaldemar | Maxxed: ${CALLERID(num)}, it's a function now. |
| 16:26.58 | Maxxed | sweet, thats the one |
| 16:27.01 | Maxxed | cool beans :) |
| 16:28.52 | *** join/#asterisk QaDeS (n=mklaus@dslb-084-056-225-094.pools.arcor-ip.net) |
| 16:30.48 | Maxxed | damnit i forgot how cool asterisk is! sheet this is awsumness times 10 ;D |
| 16:30.51 | *** join/#asterisk zeroHalo (n=zeroHalo@75.150.77.161) |
| 16:31.44 | ManxPower | Maxxed: see the UPGRADE*.txt files for info on important changes. |
| 16:35.07 | Maxxed | heck man, iv been out for a looong time now, looks like a whoooole buncha stuff has changed |
| 16:35.15 | Maxxed | but the overall stuff is coming back |
| 16:36.31 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 16:39.46 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 16:41.47 | ManxPower | read ALL the upgrade files |
| 16:43.39 | Maxxed | good advice ;) |
| 16:47.03 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 16:47.36 | fallenstorm | [Jul 8 18:45:48] WARNING[18724] chan_mgcp.c: Failed to bind to 0.0.0.0:2727: Address already in use |
| 16:47.52 | fallenstorm | in what config can i fix it? |
| 16:48.01 | fallenstorm | i cant find =\ |
| 16:49.08 | ManxPower | fallenstorm: that indicates you already have an MGCP application on the computer (server or client) or that Asterisk is already running. |
| 16:50.25 | fallenstorm | ManxPower yes, i launch 2 asterisks |
| 16:50.40 | fallenstorm | but on the different IPs |
| 16:50.51 | ManxPower | fallenstorm: Asterisk does not support that. |
| 16:51.12 | ManxPower | If you insist on doing that you won't get much help here because nobody does that. |
| 16:51.12 | fallenstorm | but could not find where port 2727 configure |
| 16:51.17 | *** join/#asterisk Mw3 (i=mw3@mw3.hu) |
| 16:51.24 | bmoraca | fallenstorm: just a huge, you know, guess here...but I think i'd probably check mgcp.conf...but, you know, that's just a guessw |
| 16:51.27 | ManxPower | fallenstorm: that's MGCP. |
| 16:51.56 | fallenstorm | yipe |
| 16:51.58 | fallenstorm | =) |
| 16:52.18 | ManxPower | fallenstorm: don't worry. If you get past this issue you still have all sorts of problems with file locking, logging, database corruption, etc. |
| 16:53.26 | ManxPower | Why are you using MGCP? |
| 16:53.48 | fallenstorm | any issue has a solve |
| 16:54.19 | ManxPower | fallenstorm: Correct. The only question is how many months will you work on this. |
| 16:54.39 | ManxPower | As I said, you're on your own. I wish you the best of luck. |
| 16:54.49 | fallenstorm | ManxPower i even dont know what is it MGCP |
| 16:54.51 | fallenstorm | =) |
| 16:55.32 | ManxPower | Asterisk has EXTENSIVE support built in for running multi-tennant and virtual PBXs. |
| 16:55.42 | ManxPower | fallenstorm: then disable MGCP |
| 16:55.48 | fallenstorm | ok |
| 16:56.18 | bmoraca | VMWare ftw for virtual PBXs |
| 16:57.08 | ariel_ | depending on how you want the setups, I would first start with different contexts, then maybe using xen but again timers and other issues come to play. |
| 16:57.46 | *** join/#asterisk The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel) |
| 16:58.01 | bmoraca | i've never had any timing issues using ESXi...i've had very good results running 7 pbxs on a single server with ESXi |
| 16:58.47 | bmoraca | i could have put more, but i ran out of disk space |
| 17:03.23 | ariel_ | bmoraca: it really depends on what each of the asterisk is doing. |
| 17:03.47 | bmoraca | full PBX for a remote office...i'm bringing one online that will have 27 phones |
| 17:04.13 | ManxPower | How many MeetMe's can you do on that VMWare box? |
| 17:04.13 | ariel_ | 27 phones, meetme's, g729? ??? |
| 17:04.16 | *** join/#asterisk jamesh1 (n=jhenders@xob.neospire.net) |
| 17:04.48 | bmoraca | haven't tested, but we've had several up at once |
| 17:05.21 | bmoraca | and no g729. though, I'm considering it. if I do 729, it'll be passthrough only on the virtual boxes, though, with encode/decode offloaded on a different box |
| 17:06.39 | ManxPower | I could never choke down the "run the pbx remote over the internet" Kool-Aid |
| 17:08.28 | bmoraca | it works great, and my customers love it...they lease the PBX time and phones from me...flat monthly rate, no capital outlay...win win |
| 17:09.37 | bmoraca | I've got one customer that has 6 locations...that's 6 phone bills and 6 PBXs to maintain...each location is small (5-10 phones), so dedicated PBX to all sites is kind of pointless...single virtual PBX, though...that's the ticket |
| 17:10.11 | ManxPower | bmoraca: I solved that problem with Frac-T1s with QoS and Cisco Routers |
| 17:10.17 | *** join/#asterisk phurl (n=mdupont@82.114.94.9) |
| 17:10.39 | bmoraca | cisco routers are expensive...virtual PBX costs me nothing and I get to charge them $1500/mo |
| 17:10.56 | ManxPower | bmoraca: They paid $400 each for the routers. |
| 17:11.31 | ManxPower | bmoraca: Do you also get to charge them when there are issues somewhere between the server and the phones? |
| 17:12.05 | bmoraca | ManxPower: hasn't happened yet, but they sign a ToS which says we're not responsible for issues with the interwebs |
| 17:12.07 | ariel_ | wow that is allot of money |
| 17:12.39 | ariel_ | 6 location really can be run off one asterisk with the correct context setup in any case. |
| 17:12.51 | *** join/#asterisk jplank (n=GBove@cpe-098-024-169-236.carolina.res.rr.com) |
| 17:13.34 | ManxPower | ariel_: I agree with that. I guess my customers were just assholes compared to bmoraca's customers. I used to get a phone call if a single fax didn't go thru or if a single phone call had audio problems. |
| 17:13.36 | bmoraca | ariel_: their previous phone bill for all 6 locations was upwards of $1800...so they save $300/mo and get a brand new phone system...win for them...and win for me, because it costs me about $100/mo to actually run their service (includes colocation, PRI channels, and data charges) |
| 17:13.41 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 17:15.09 | bmoraca | ariel_: these 6 locations, being that they're all the same customer, are a single virtual PBX...but i'm not going to put more than one customer on the same PBX...too many problems to worry about, and these servers are cheap |
| 17:15.46 | *** join/#asterisk Davidf88 (i=david@proxima.lp0.eu) |
| 17:15.51 | bmoraca | used ProLiant DL380 G3s ftw! |
| 17:15.56 | ariel_ | I am actually moving off the hosted pbx setups, I find it easier to setup small asterisk pbx on each site. |
| 17:17.07 | ariel_ | at the remote ends I have been using some of these type of boxes: under $ 500 dollars and there great actual pbx's |
| 17:17.11 | ariel_ | http://www.rowetel.com/ucasterisk/index.html |
| 17:17.54 | jpsharp | Ooh, those are neat looking. |
| 17:18.04 | bmoraca | ariel_: i thought about that...but then I realized that with an office with 6 phones, it's really kind of pointless, and the less equipment at the customer's site, the better...though that box is kind of neat |
| 17:18.12 | jplank | if dadhi_cfg -vv looks correct, and the files dahdi_genconf generate look correct, what would make a dahdi show channels not show the channels? |
| 17:18.36 | ariel_ | no channels up and running |
| 17:18.46 | jplank | what do you mean by that? |
| 17:19.04 | tzafrir_laptop | jplank, asterisk not started? |
| 17:19.16 | jplank | I'm doing the dahdi show status from the asterisk cli |
| 17:19.31 | jplank | dmesg looks to load all 24 channel (this is a tdm2400p) |
| 17:19.33 | Zuchmir2 | how do i make sure * in the audio path |
| 17:19.47 | ManxPower | Zuchmir2: cantrinvite=no in sip.conf |
| 17:19.50 | jpsharp | make sure canreinvite=no |
| 17:20.02 | tzafrir_laptop | jplank, what's the output of lsdahdi |
| 17:20.15 | bmoraca | Zuchmir2: additionally, if you specify any options in your Dial() statement, it will force asterisk to be in the audio path |
| 17:20.44 | jplank | chan 1 red, 2-7 not, 8-24 red |
| 17:20.50 | jplank | 1 FXO FXSKS (SWEC: MG2) RED |
| 17:23.53 | *** join/#asterisk fallenstorm (n=fallen@89-178-9-13.broadband.corbina.ru) |
| 17:24.00 | jplank | could it have anything to do with how the lines are xconnected? |
| 17:27.01 | jpsharp | red alarm says its not seeing the line. |
| 17:28.46 | jplank | but even in red alarm, should I still see the channels in dahdi show channels ? |
| 17:29.57 | ManxPower | jplank: yes. |
| 17:30.21 | ManxPower | you have problems with your ASTERISK dahdi config, not the KERNEL dahdi setup. |
| 17:30.43 | jplank | I agree |
| 17:30.51 | jplank | but I can't find where its messed up |
| 17:31.02 | jplank | genconf works |
| 17:31.07 | jplank | system.conf looks good |
| 17:31.26 | jplank | dahdi-channels.conf looks good |
| 17:31.50 | jplank | and chan-dahdi.conf includes dahdi-channels.conf |
| 17:31.59 | jplank | I'm sure I'm missing something, but I cant seem to find it |
| 17:32.27 | andres833 | hi |
| 17:32.35 | ManxPower | jplank: configure it by hand. I've never actually had any of the zaptel auto config work for me. |
| 17:32.45 | ManxPower | it always seems to set it up for PRI or something stupid like that. |
| 17:33.01 | andres833 | what is this /etc/init.d/asterisk start |
| 17:33.01 | andres833 | Starting Asterisk PBX: Asterisk already running on /var/run/asterisk.ctl. Use 'asterisk -r' to connect. |
| 17:33.09 | tzafrir_laptop | jplank, grep channel /etc/asterisk/*dahdi*.conf |
| 17:33.34 | jplank | all 24 are in there tzafrir_laptop |
| 17:33.34 | ManxPower | andres833: that message is correct. Asterisk is already running. |
| 17:33.44 | tzafrir_laptop | andres833, hmmm... why not try that? |
| 17:33.45 | ManxPower | jplank: pastebin the files for people to look at |
| 17:36.43 | andres833 | ManxPower, mira http://paste.debian.net/41359/ |
| 17:37.02 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 17:37.06 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 17:37.51 | jplank | here is dahdi-channels.conf, chan-dahdi.conf and system.conf http://pastebin.ca/1488488 |
| 17:37.59 | *** join/#asterisk bsilberman (n=bsilberm@65.213.221.252) |
| 17:37.59 | jplank | let me know if I'm missing anything |
| 17:39.16 | Katty | grooves. |
| 17:39.23 | *** join/#asterisk heison (n=heison@204.29.161.10) |
| 17:39.35 | Katty | so stuffed. |
| 17:39.43 | Katty | ate too much for lunch, but it was soooooooo sooo good |
| 17:39.55 | tzafrir_laptop | jplank, chan_dahdi.conf (with a _) right? |
| 17:40.00 | ManxPower | andres833: stop. you connect to Asterisk by using "asterisk -r" not "asterisk start" |
| 17:40.18 | Katty | sauted turkey breast, on whole wheat bagel, with red onion, green pepper, and cucumber. also sauted sugar snap peas and a baked apple for dessert. |
| 17:40.52 | ManxPower | andres833: What EXACTLY are you trying to do? |
| 17:41.07 | Katty | ManxPower: he's trying to do what he tries to do every night |
| 17:41.10 | Katty | ManxPower: TRY TO TAKE OVER THE WORLD |
| 17:41.19 | tzafrir_laptop | jplank, 'chan*-*dahdi.conf' is not the right name |
| 17:41.27 | andres833 | ManxPower, but i need that the asterisk server run when the server start |
| 17:41.37 | Katty | andres833: run levels. |
| 17:41.43 | ManxPower | andres833: "killall -9 asterisk" |
| 17:41.45 | Katty | andres833: add it to your rc.local |
| 17:41.48 | ManxPower | then try asterisk start |
| 17:42.02 | tzafrir_laptop | rc.local ? |
| 17:42.02 | tzafrir_laptop | that is wrong |
| 17:42.03 | jplank | i'm an idiot |
| 17:42.09 | jplank | thanks tzafrir_laptop |
| 17:42.27 | Katty | tzafrir_laptop: i couldn't swore that's what i added it to. |
| 17:42.44 | Katty | tzafrir_laptop: however it's been ahwile, and my blog o references is not working due to dydns issues |
| 17:43.21 | tzafrir_laptop | It will get Asterisk started at boot, but that's not the proper way |
| 17:43.43 | tzafrir_laptop | if you have a init.d script, just use standard runlevel scripts |
| 17:43.58 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 17:44.06 | tzafrir_laptop | Otherwise you'll never get to the "take over the world" part |
| 17:44.07 | jplank | tzafrir_laptop: I don't think thats right, I have another box running it as chan_dahdi.conf |
| 17:44.35 | Katty | hmm |
| 17:44.42 | Katty | it appears i just have mount commands in rc.local |
| 17:45.10 | jplank | tzafrir_laptop: it is a underscore, not a dash |
| 17:45.35 | tzafrir_laptop | Katty, and you don't use fstab for that because? |
| 17:45.39 | *** join/#asterisk securevoip (n=securevo@173-15-197-73-BusName-Richmond.hfc.comcastbusiness.net) |
| 17:46.04 | Katty | tzafrir_laptop: because i enjoy hearing all your negative comments. |
| 17:46.57 | tzafrir_laptop | jplank, is there a 'dahdi' command on asterisk? |
| 17:47.01 | jplank | yes |
| 17:47.13 | tzafrir_laptop | 'dahdi show channels' shows anything? |
| 17:47.15 | jplank | I see the pseudo channel on a dhadi show channels |
| 17:47.15 | *** part/#asterisk tobias (n=tobias@71.70.219.40) |
| 17:47.17 | Katty | oh here we go |
| 17:47.22 | Katty | /etc/init.d/asterisk |
| 17:47.32 | Katty | on and then symlinks to rc2.d |
| 17:48.08 | Katty | i have FoP and isymphony in rc.local tho |
| 17:49.38 | bmoraca | ewww...FoP |
| 17:49.40 | jplank | I have to be missing something |
| 17:49.52 | bmoraca | jplank: did you try restarting Asterisk? |
| 17:49.56 | bmoraca | restart now |
| 17:50.00 | jplank | yea |
| 17:50.03 | Katty | bmoraca: it's neat to tinker with |
| 17:50.07 | jplank | also a reboot |
| 17:50.14 | Zuchmir2 | what ports have to be open on fw to allow sip to work properly? (I do register w/VSP) |
| 17:50.22 | jplank | and I've stopped asterisk, restarted dahdi and then started back up |
| 17:50.39 | Katty | there sure is a lot of negativity in here today |
| 17:50.45 | ManxPower | Zuchmir2: port 5060 UDP and ports 10,000-20,000 UDP (default) or whatever ports you specify in /etc/asterisk/rtp.conf |
| 17:51.03 | ManxPower | Each call uses two UDP ports for audio and UDP port 5060 for signaling |
| 17:51.39 | bmoraca | Zuchmir2: if you're registering with a provider, you don't need to forward anything (just make sure your router is properly configured and you use nat=yes in the user context). for phones registering to your asterisk, you'll need to forward 5060 and the RTP ports referenced above |
| 17:52.09 | bmoraca | although, technically, i've never bothered with the nat=yes |
| 17:52.13 | ManxPower | bmoraca: he did say "fw" |
| 17:52.53 | bmoraca | ManxPower: outbound connections don't need any firewall policy exemptions |
| 17:53.01 | ManxPower | Hopefully he's not an idiot and said "fw" when he meant "nat" |
| 17:53.08 | bmoraca | although, I imagine he's just talking about a NAT router |
| 17:53.21 | ManxPower | bmoraca: I dunno about your firewall, but my firewalls don't let ANYTHING thru that I don't specify |
| 17:53.43 | ManxPower | inbound or outbound |
| 17:54.07 | *** join/#asterisk QaDeS (n=mklaus@dslb-084-056-225-094.pools.arcor-ip.net) |
| 17:54.27 | *** join/#asterisk errotan (n=errotan@5403E64C.catv.pool.telekom.hu) |
| 17:54.59 | Zuchmir2 | hmm, i have 2 VSPs, one the voice goes ok, the other gives trouble :-( |
| 17:55.30 | bmoraca | Zuchmir2: could be many issues there |
| 17:55.59 | ManxPower | Zuchmir2: Do you have a firewall or just a NAT router? |
| 17:56.48 | Zuchmir2 | fw |
| 17:56.57 | bmoraca | what model? |
| 17:57.05 | securevoip | Is one of the VSPs ipcomms.net by chance? |
| 17:57.07 | ManxPower | So you are not forwarding any ports or anything like that. |
| 17:57.45 | Zuchmir2 | 1 VSP requires registration (that's the one that has problems), the other does NOT use registration (that one the voice is OK) |
| 17:57.54 | Zuchmir2 | juniper |
| 17:58.13 | ManxPower | Zuchmir2: turn OFF any ALG or SIP support in the firewall. |
| 17:58.23 | ManxPower | Let Asterisk do it by using nat=yes |
| 17:58.27 | bmoraca | i believe they have an inspect or ALG for SIP, similar to Cisco's fixup...you need to turn that OFF |
| 17:58.30 | Zuchmir2 | i tried it with as well as without |
| 17:58.41 | ManxPower | bmoraca: Hey! We agree on something! *grin* |
| 17:58.46 | bmoraca | lol |
| 17:58.55 | jplank | grrr no one seen a typo in my config files or anything like that? |
| 17:59.05 | ManxPower | Zuchmir2: it won't work correctly if you have that enabled on your FW |
| 17:59.08 | bmoraca | i was kind of hoping he'd say Sonicwall, so I could yell at him and tell him how much they suck |
| 17:59.14 | jplank | is asterisk is seeing the pseudo channel, doesn't that mean *something* is programmed correctly? |
| 17:59.19 | jplank | if asterisk* |
| 17:59.32 | ManxPower | Asterisk nat=yes + Firewall ALG = doesn't work. |
| 17:59.49 | ManxPower | jplank: yes, it means Asterisk sees DAHDI, but does not have a config or any of the channels. |
| 18:00.03 | jplank | what could I be missing? |
| 18:00.11 | ManxPower | why don't you just try using ONE chan_dhadi config file rather than several that you are #Including |
| 18:00.27 | jplank | triyng that now |
| 18:00.28 | Zuchmir2 | how about nat=no + ALG? |
| 18:00.36 | ManxPower | Zuchmir2: that may or may not work. |
| 18:00.47 | bmoraca | never had any luck with ALGs |
| 18:00.59 | ManxPower | Ragardless we can't help you if your firewall is messing with SIP packets |
| 18:01.04 | bmoraca | most interesting i've ever used was Adtran's transparent SIP proxy...really, really weird |
| 18:01.42 | bmoraca | but it worked really well |
| 18:01.49 | jplank | something like http://pastebin.ca/1488510 should work right? |
| 18:02.32 | *** part/#asterisk sack (n=sack@132.Red-88-24-153.staticIP.rima-tde.net) |
| 18:03.34 | *** join/#asterisk Laureano (n=Laureano@190.245.101.140) |
| 18:03.34 | Zuchmir2 | turned off ALG, did not help |
| 18:03.39 | ManxPower | jplank: do this. unload chan_dahdi.so then load chan_dahdi.so see if you see any errors |
| 18:03.45 | ManxPower | Zuchmir2: leave it turned off. |
| 18:04.16 | securevoip | Zuchmir2: Who is the VSP that you are having problems with? |
| 18:04.21 | ManxPower | jplank: yes, but everything after the last channel => line is ignored |
| 18:04.28 | Zuchmir2 | phonepower |
| 18:05.31 | bmoraca | Zuchmir2: enable sip debug and see what's going on at the packet level |
| 18:05.39 | jplank | ManxPower: no errors |
| 18:05.59 | jplank | hmmm |
| 18:06.05 | jplank | no errors, but look at this |
| 18:06.28 | jplank | http://pastebin.ca/1488520 |
| 18:07.05 | bmoraca | what's in your zapata.conf files? |
| 18:07.17 | jplank | haaaa |
| 18:07.18 | jplank | got it |
| 18:07.21 | jplank | weird though |
| 18:07.39 | jplank | why would dahdichanname=no keep it from loading the channels? |
| 18:07.58 | jplank | I thought all that did was allow ZAP to be used in addition to DAHDI? |
| 18:08.46 | jplank | I'm using it in another setup, looks to be the same config (more or less) and it works |
| 18:10.01 | ManxPower | jplank: notice it loaded the zaptel configs not the dahdi configs in your pastebin |
| 18:11.10 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 18:15.15 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 18:15.27 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 18:16.47 | bsilberman | question -- I have a Sangoma A108 card in my linux box, which is plugged into a Cisco ws-x6608-t1 card in a 6509 switch direct connect. the card is being controlled by card manager 4.1 |
| 18:17.20 | bsilberman | I have yet to test successfully between the two. Every time we call, we get no connection, or when we do connect, we hear a buzzing sound |
| 18:17.42 | bsilberman | we have tried E&M, E&M Wink, and ISDN QSIG on the line |
| 18:17.54 | bsilberman | anyone have any idea or have run into this previously? |
| 18:18.25 | *** join/#asterisk knipster (n=knipster@164.55.254.106) |
| 18:18.41 | *** join/#asterisk bgerlich (n=bgerlich@81-186-237-54.cityconnect.pl) |
| 18:23.21 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 18:23.58 | [TK]D-Fender | bsilberman: What protocol is the Cisco using? Who's CPE? |
| 18:25.22 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:29.28 | bsilberman | astersk is CPE, and re protocol on the cicso, you want to know whats on the t1? |
| 18:29.53 | bsilberman | over the t1 is esf,b8zs |
| 18:29.56 | ariel_ | Cisco should be Net and setup as a PRI |
| 18:30.01 | bsilberman | and e&m wink |
| 18:30.45 | *** join/#asterisk ramindia (n=balajibh@202.63.96.10) |
| 18:30.49 | ariel_ | why would you want to use e&m wink when PRI digital would be far better and give you more info |
| 18:30.55 | bsilberman | tried pri, still got the buzzing noize |
| 18:31.16 | bsilberman | it would, if it worked |
| 18:31.36 | bsilberman | regardless of the pri or plain t1, still getting the buzzing noise |
| 18:32.00 | ariel_ | buzzing noise can be due to configuration, bad cable between them, the list is really long |
| 18:32.34 | ariel_ | in pri mode do you get your layer 3 up (d channel)? |
| 18:33.01 | ramindia | [TK]D-Fender: hi |
| 18:33.14 | ariel_ | have you ever made a loop back cable bsilberman ? |
| 18:33.33 | bsilberman | cable was replaced, port on cisco was moved to a new card |
| 18:34.01 | bsilberman | astersk card is still plugged into port 1, has not changed |
| 18:34.24 | bsilberman | i have never made a t1 loop back cable, no |
| 18:34.29 | ramindia | hey i have some typical question...iam calling from asterisk to calling card.. and enter pin and called DID.. can i get C tone from the DID |
| 18:34.34 | bsilberman | what port to where? |
| 18:35.13 | [TK]D-Fender | ramindia: "C tone"? |
| 18:35.53 | ariel_ | I have allot of asterisk digium boards connected to Cisco's mostly via E1 since I can get more channels that way. But either case I have not set them up on a 6509 mostly on 2821, 36XX and 53XX series |
| 18:36.42 | bsilberman | no stats on cisco re a bad cable,... no typical error counters suggesting a bad cable |
| 18:37.27 | bsilberman | so, assuming that the cisco itself is ok, and that the cable is ok, all that is left is the asterisk card, and configs on both |
| 18:37.28 | [TK]D-Fender | bsilberman: Aside from bad noise do you get the conversation at all? Any errors generated? |
| 18:37.40 | bsilberman | no conversation, only a steady tone |
| 18:38.01 | [TK]D-Fender | bsilberman: Seriously avoid E&M |
| 18:38.09 | [TK]D-Fender | bsilberman: and pastebin your configs |
| 18:38.14 | ramindia | Connect Tone |
| 18:38.43 | [TK]D-Fender | ramindia: * does not care about secondary tones from them |
| 18:39.14 | ramindia | the DID also mapped in the same * box |
| 18:39.22 | ramindia | so call is in loop |
| 18:39.50 | bsilberman | ok, now config is using national pri_cpe |
| 18:39.53 | [TK]D-Fender | ramindia: And the point of calling yourself through a calling card is...? |
| 18:39.54 | ramindia | My goal is to find out success of calling card and PIN |
| 18:40.18 | [TK]D-Fender | ramindia: I've answered this question several times. NOT GOING TO HAPPEN. |
| 18:40.20 | ramindia | with that PIN i can make calls or not |
| 18:40.37 | ramindia | yes u are told me |
| 18:41.10 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:41.14 | ramindia | any other method to success this situation.. since it need to be deploy my test system..who give less rate for this calling card so i want to use this calling cards |
| 18:41.19 | [TK]D-Fender | ramindia: So how many more times are you going to ask the same pointless question? |
| 18:41.35 | bsilberman | from /etc/asterisk/chan_dahdi.conf: ;Sangoma A108 port 1 [slot:4 bus:13 span:1] <wanpipe1> |
| 18:41.35 | bsilberman | switchtype=national |
| 18:41.35 | bsilberman | context=incoming |
| 18:41.35 | bsilberman | group=1 |
| 18:41.35 | bsilberman | echocancel=yes |
| 18:41.36 | bsilberman | signalling=pri_cpe |
| 18:41.41 | [TK]D-Fender | ramindia: * has not functionality for this. Go invent your own dial command. |
| 18:42.17 | bsilberman | from /etc/dahdi/system.conf: #Sangoma A108 port 1 [slot:4 bus:13 span:1] <wanpipe1> |
| 18:42.17 | bsilberman | span=1,0,0,esf,b8zs |
| 18:42.17 | bsilberman | bchan=1-23 |
| 18:42.17 | bsilberman | echocanceller=mg2,1-23 |
| 18:42.17 | bsilberman | hardhdlc=24 |
| 18:43.06 | bsilberman | versions: asterisk-1.4.24 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.10 wanpipe-3.4.1 |
| 18:43.22 | *** join/#asterisk pierrelux (n=pierrelu@IP-208-88-110-46.mtl.fibrenoire.ca) |
| 18:44.42 | [TK]D-Fender | bsilberman: PASTEBIN |
| 18:44.49 | [TK]D-Fender | bsilberman: Stop spamming.. |
| 18:44.51 | [TK]D-Fender | ~pb |
| 18:44.52 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/, or apt-get install pastebinit |
| 18:45.36 | [TK]D-Fender | bsdYou should also specify your dchan=24 |
| 18:45.48 | pierrelux | let say my client uses only speex 32 kHz and asterisk is configured with speex 8 kHz and some other codecs. Is it normal that asterisk returns speex 8000 after SDP negotiation ? Is is the way the RFC says to do so ? It's up to the application/device to resample, right ? |
| 18:46.20 | [TK]D-Fender | bsilberman: Also if your Cisco is NET, your span should be 1,1,0 |
| 18:46.40 | tzafrir_laptop | [TK]D-Fender, are you sure? IIRC sangoma cards use hardhdlc |
| 18:46.58 | tzafrir_laptop | err... the drivers |
| 18:47.01 | [TK]D-Fender | tzafrir_laptop: No, normal dchan. |
| 18:47.05 | bsilberman | http://pastebin.com/me2e4b1f /etc/asterisk/system.conf |
| 18:47.23 | tzafrir_laptop | If they do, they probably use a patched dahdi |
| 18:47.47 | tzafrir_laptop | but I don't really know that |
| 18:47.53 | bsilberman | http://pastebin.com/mdec0590 is the /etc/asterisk/chan_dahdi.conf |
| 18:48.13 | bsilberman | the other one was /etc/dahdi not /etc/asterisk |
| 18:48.45 | *** join/#asterisk mweichert (n=mweicher@216.13.154.21) |
| 18:54.18 | jaytee | trying to setup a blacklist using just the function so it will be 1.6 compliant since Lookupblacklist is deprecated. if i use for example, exten = _XXXX,1,GotoIf(BLACKLIST()?thiscontext,_XXXX,blacklisted) and the callerid matches an entry in the AstDB that should jump to the priority labeled blacklisted and if no match continue on the next priority? |
| 18:55.49 | beek | jaytee: Yep. |
| 18:57.17 | jaytee | wasn't sure if I had to use Set with BLACKLIST to set a var to what it returns and then evalutate in GotoIf. I'd seen an example earlier when surfing but couldn't find it again today so I was going by memory (which at my age I don't rely heavily on) |
| 18:57.36 | jaytee | s/evalutate/evaluate |
| 18:57.41 | beek | The function returns 1 or 0 |
| 18:58.37 | jaytee | yeah, but I wasn't sure if I needed a $ with braces around the BLACKLIST function or not. I think the example I saw was just as I typed it. |
| 18:59.54 | *** join/#asterisk lasko (n=lasko@70.102.15.210) |
| 19:00.25 | lasko | Has anyone been able to get the schedule=yes function with Meetme in 1.6? |
| 19:00.31 | lasko | to work |
| 19:03.17 | beek | jaytee: I just tested this against 1.6.0.10: exten => 1670,n,GotoIf(${BLACKLIST()}?hangup); |
| 19:03.22 | beek | Worked fine. |
| 19:04.10 | ManxPower | jaytee: You can't Goto* a PATTERN, you must goto actual digits |
| 19:04.24 | [TK]D-Fender | Useless function, just use DB() yourself |
| 19:04.46 | beek | [TK]D-Fender: But its in there and certainly shorter. |
| 19:05.00 | [TK]D-Fender | beek: and deprecated |
| 19:06.08 | beek | They need to note that in 'core show function' output... |
| 19:06.18 | beek | Or is that as of 1.6.1 series? |
| 19:06.25 | jaytee | the blacklist function isn't deprecated, just the app Lookupblacklist |
| 19:06.45 | [TK]D-Fender | jaytee: Either way :) |
| 19:07.03 | jaytee | ManxPower, and thanks for pointing out the PATTERN match part of the GotoIf. |
| 19:07.44 | beek | Anyone here having trouble compiling wanpipe 3.5.4 against dahdi-linux 2.2.0? |
| 19:07.49 | jaytee | I'll just point it to a unique exten instead |
| 19:08.46 | jaytee | and thanks also beek |
| 19:09.13 | jaytee | looks like I need to enclose the function with ${ } |
| 19:12.34 | ManxPower | jaytee: You can go to context,${EXTEN},1 as well |
| 19:15.05 | jaytee | [TK]D-Fender, unless the BLACKLIST() function is being deprecated which I don't believe it is then it's less code than using DB(). |
| 19:15.16 | beek | and more obvious |
| 19:15.59 | jaytee | [TK]D-Fender, I'll probably never understand all this as thoroughly as you but then I'm also lazy and work with what I have instead of writing my own Pascal style parser :-) |
| 19:16.17 | ManxPower | jaytee: everything that's being deprecated is listed in UPGRADE*.txt |
| 19:16.34 | [TK]D-Fender | jaytee: You're just jealous because I had way too much free time on my hands ;) |
| 19:16.37 | *** join/#asterisk scarecrowed (n=fabiano@200.103.106.38) |
| 19:17.28 | jaytee | ManxPower, I looked and it also says it for Lookupblacklist in the book and at the CLI but nothing in 1.6.0.x upgrade texts says anything about the BLACKLIST function being dep'd in 1.6 |
| 19:17.57 | *** join/#asterisk tfrew (n=tfrew@75-145-244-108-Richmond.hfc.comcastbusiness.net) |
| 19:18.15 | jaytee | [TK]D-Fender, yes, I'm jealous because of that, because your depth of expertise far exceeds mine and last but not least because you still have a full head of hair |
| 19:19.02 | beek | hair is overrated. |
| 19:19.06 | ISO9001 | ^^^ |
| 19:19.27 | beek | what I want to know is why those who have full heads of hair insist on shaving it down to damned near bald? |
| 19:20.11 | tfrew | i have near afro length hair |
| 19:21.03 | ISO9001 | Personally I prefer to flip flop from afro -> shaved head. |
| 19:21.10 | ISO9001 | Take THAT, bald people. |
| 19:21.38 | jaytee | I want have a ponytail and wear red flannel shirts, jeans and sandals like the old unix gurus at Berkely |
| 19:22.06 | ManxPower | I look like I'm on chemo when I shave my head. |
| 19:22.25 | beek | If I let it grow I could get a ponytail out of the hair growing from my ears. |
| 19:22.55 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 19:31.08 | jaytee | I was going to try the ear hair combover to get hairdo like The Donald |
| 19:31.59 | jaytee | it's just proof that if there is a God he's a sadist cuz hair stops growing where you want to keep it and starts growning where you don't want it. |
| 19:35.45 | *** part/#asterisk knipster (n=knipster@164.55.254.106) |
| 19:35.52 | bmoraca | jaytee: so you like the little boy look, then? |
| 19:36.03 | tfrew | jaytee: epic |
| 19:36.12 | [TK]D-Fender | jaytee: Proof that God has a sense of humour : the platypus |
| 19:36.26 | Qwell | babelfish - just sayin' |
| 19:40.21 | Qwell | (I would be very sad if nobody got that btw) |
| 19:41.40 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 19:43.43 | *** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130) |
| 19:44.49 | *** join/#asterisk ldsjohn (n=jbunn@c-98-246-83-83.hsd1.wa.comcast.net) |
| 19:45.13 | ldsjohn | is it normal for asterisk to create huge files in tmp? they are like 6 gigs a peice and all are astmail-gibberish |
| 19:45.26 | ManxPower | <-- has the HHGTTG book, audio book, BBC Radio series, BBV TV mini series, and both movies. |
| 19:45.55 | ManxPower | ldsjohn: Asterisk does not by default create anything in /tmp IIRC. |
| 19:46.09 | ManxPower | Maybe you are using AGI or other scripts? |
| 19:47.58 | jaytee | "hmmm, I've got all these spare parts left over...what could I make?..... I know! an egg laying mammal with poisonous hind claws and a duckbill" |
| 19:48.00 | ldsjohn | well I have a box that started as pbx in a flash, but I ripped out the configs and killed the amportal part, then compiled asterisk 1.4 and then built my own configs. its a server used just for voicemail, and I am using mysql to configure the mailboxes. about 1200 mailboxes |
| 19:48.43 | ldsjohn | im not using any agi scripts, and the manager part is locked down so none could be running, every couple weeks the box runs out of space. and there are 12 gigs in 2 files each about 6 gigs a peice in /tmp |
| 19:48.59 | ldsjohn | they are astmail-YSDF and astmail-LKJOU ( the letters are different every time ) |
| 19:49.27 | ldsjohn | mysql stops working and nobody can leave messages, and then I delete those 2 files and restart asterisk and everything starts working again |
| 19:49.31 | ManxPower | ldsjohn: and yet asterisk does not send mail. It just hands it off to the local sendmail or compat |
| 19:49.58 | jaytee | [TK]D-Fender, and somedays my job feels like being trapped in a room being forced to listen to Vogon poetry :-) |
| 19:49.59 | ManxPower | looks like you need to start looking at things using lsof to see what process is creating those files. |
| 19:50.04 | ldsjohn | hrm this asterisk install isn't sending mail. could it be that it is trying to send voicemail emails. and they are just loading into a temp folder? |
| 19:50.31 | ManxPower | ldsjohn: I would have to look at the source code to see. |
| 19:50.52 | ManxPower | pastebin your voicemail.conf |
| 19:51.23 | [TK]D-Fender | goes to refill his brownian substance vessel |
| 19:51.53 | ldsjohn | my voiecemail config just has one line operator=yes in [general] |
| 19:52.24 | ldsjohn | everything else is in mysql |
| 19:54.38 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:57.05 | bmoraca | ldsjohn: have you looked at your mail.log file to see if maybe the mail is being hung up in the MTA? |
| 20:00.38 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 20:02.01 | *** join/#asterisk Joel (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 20:02.23 | Joel | what level do you have to set for messages to get agi debugging to log there? |
| 20:02.31 | WindowsUser | hrm, does the spa3102 send pstn calls to the handset by default? |
| 20:02.34 | WindowsUser | Joel: try 3 |
| 20:02.57 | WindowsUser | oh, to log where? a file? or to the console? |
| 20:03.22 | Joel | WindowsUser a file, "messages" |
| 20:04.40 | Joel | full => notice,warning,error,debug,verbose is for example a logger line I have running, which logs to /var/log/asterisk/full |
| 20:04.44 | Joel | but I see no agi debug output |
| 20:06.54 | [TK]D-Fender | WindowsUser: I doubt anything is 'default" |
| 20:07.32 | WindowsUser | aye |
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| 20:12.31 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-c73cf729c03545e8) |
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| 20:21.54 | jaytee | ManxPower, beek, [TK]D-Fender, got the blacklisting feature working great. thanks for all your help |
| 20:22.09 | *** join/#asterisk telecos (n=sergio@183.166.219.87.dynamic.jazztel.es) |
| 20:25.41 | [TK]D-Fender | checkout time, later all |
| 20:26.33 | *** join/#asterisk andres833 (n=andres83@190.144.102.122) |
| 20:29.28 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.108.111) |
| 20:29.30 | DelphiWorld | hi |
| 20:29.39 | DelphiWorld | what is the default asterisk AMI user / password? |
| 20:29.59 | ManxPower | DelphiWorld: there are no default passwords in Asterisk |
| 20:30.41 | DelphiWorld | ManxPower: and user? |
| 20:31.01 | ManxPower | Any passwords / users are either configured by you or set up by whatever GUI you are running. |
| 20:31.11 | ManxPower | manager stuff is configured in /etc/asterisk/manager.conf |
| 20:31.30 | DelphiWorld | ManxPower: ok i will try |
| 20:32.03 | ManxPower | Of course calling a GUIfied Asterisk "Astersisk" is like saying you are running DOS when you are running Windows 95. Yes, Windows 95 runs on top of DOS. |
| 20:32.35 | Nugget | ManxPower is just a Windows for Workgroups enthusiast. |
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| 20:44.42 | jaytee | Microcrap Winders |
| 20:45.07 | ManxPower | The only think Microsoft might ever make that didn't suck are vacuum cleaners. |
| 20:46.15 | jaytee | been having fun this week dealing with Sharepoint authentication problems caused by pushing Office2007 out to client computers using MS SCCM 2007. what a PITA! |
| 20:55.14 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.30) |
| 20:57.04 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 20:57.20 | ManxPower | I hope I never have to stoop so low as to have to support Microsoft stuff. |
| 20:57.38 | ManxPower | I'd do a career change before I got that desperate. |
| 20:58.06 | bmoraca | there's a lot of money in supporting Active Directory, and not because it's a piece of crap but because everyone uses it |
| 20:58.46 | hardwire | sigh. |
| 20:58.57 | ManxPower | There's a lot of money smuggling drugs too. |
| 20:59.16 | bmoraca | yes, but supporting Microsoft domains is much easier |
| 20:59.26 | ManxPower | says you. 8-) |
| 20:59.40 | *** part/#asterisk zeroHalo (n=zeroHalo@75.150.77.161) |
| 21:00.37 | bmoraca | our most profitable jobs are fixing other "tech"s' screwups |
| 21:03.20 | jaytee | quittin time. back later |
| 21:07.01 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
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| 21:31.38 | *** join/#asterisk batphone (n=wclayton@66.219.32.14) |
| 21:31.39 | batphone | can anyone recommend a CLI utility to assess VoIP call quality issues? |
| 21:31.41 | batphone | im dealing with packet8 right now and am having to set up a huge lab |
| 21:31.44 | batphone | and i need some tools to get the ball rolling |
| 21:32.41 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 21:41.54 | WindowsUser | sipp? |
| 21:46.59 | *** join/#asterisk LakeSolon (n=blake@72.11.109.214) |
| 21:55.08 | batphone | WindowsUser: thanks |
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| 22:11.07 | *** join/#asterisk citywok (n=chatzill@corpnet.csgopenline.com) |
| 22:13.31 | citywok | I am looking for a way, on either conference start, or conference end, to tell what conference number was used in meetme. It's not logged in CDR's, and it's not a channel variable on the Zap channel. any ideas? |
| 22:14.50 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
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| 22:55.48 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 23:00.32 | *** join/#asterisk Hydrant (n=aj@74.198.8.70) |
| 23:00.51 | Hydrant | hello all... is it possible to change the key that DISA wants from # to something else ? |
| 23:04.39 | *** join/#asterisk jkroon (n=jkroon@dsl-240-167-45.telkomadsl.co.za) |
| 23:05.02 | ManxPower | Hydrant: If it's not mentioned in "core show application DISA" then I doubt you can do it without changing the source code. However, it would be fairly easy to write a DISA-like app just using the dialplan. |
| 23:05.20 | jkroon | https://issues.asterisk.org/view.php?id=1574 <-- very old issue, however, i suspect I'm seeing similar issues with trunk=yes, trunktimestamps=yes and forcejitterbuffer on the IAX/2 side. |
| 23:06.40 | Hydrant | ManxPower: that's too bad... my iphone won't allow # to be programmed in for prefix dialing :-S |
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| 23:09.18 | *** join/#asterisk techie (n=techie@adsl-76-240-176-52.dsl.lsan03.sbcglobal.net) |
| 23:11.01 | *** part/#asterisk techie (n=techie@adsl-76-240-176-52.dsl.lsan03.sbcglobal.net) |
| 23:11.06 | *** join/#asterisk techie (n=techie@adsl-76-240-176-52.dsl.lsan03.sbcglobal.net) |
| 23:11.10 | *** part/#asterisk techie (n=techie@adsl-76-240-176-52.dsl.lsan03.sbcglobal.net) |
| 23:12.23 | *** join/#asterisk techie (n=techie@adsl-76-240-176-52.dsl.lsan03.sbcglobal.net) |
| 23:13.29 | *** part/#asterisk techie (n=techie@adsl-76-240-176-52.dsl.lsan03.sbcglobal.net) |
| 23:14.29 | [TK]D-Fender | Hydrant: "Key that DIS wants"? huh? |
| 23:14.37 | *** join/#asterisk techie (n=techie@adsl-76-240-176-52.dsl.lsan03.sbcglobal.net) |
| 23:14.53 | *** part/#asterisk techie (n=techie@adsl-76-240-176-52.dsl.lsan03.sbcglobal.net) |
| 23:15.44 | Hydrant | [TK]D-Fender: DISA wants the PIN followed by # |
| 23:16.01 | *** join/#asterisk techie (n=techie@adsl-76-240-176-52.dsl.lsan03.sbcglobal.net) |
| 23:16.05 | Hydrant | Unfortunately, I can't put a # on my phone |
| 23:16.30 | Hydrant | so I need to see if I change that # to a *... or better, just a pause |
| 23:16.46 | *** part/#asterisk techie (n=techie@adsl-76-240-176-52.dsl.lsan03.sbcglobal.net) |
| 23:16.51 | *** join/#asterisk techie (n=techie@adsl-76-240-176-52.dsl.lsan03.sbcglobal.net) |
| 23:18.56 | ManxPower | sounds like you need a new phone |
| 23:20.09 | *** part/#asterisk hamush (n=hamush@24-155-177-34.dyn.grandenetworks.net) |
| 23:24.17 | [TK]D-Fender | Hydrant: Sounds like you shoul do your OWN auth to enter DISA and not us ITS one <- |
| 23:24.21 | [TK]D-Fender | use* |
| 23:26.53 | jkroon | ok, from what i've now read in the iax/2 protocol specification it only makes sense to set trunktimestamps globally in iax.conf or not at all ... it doesn't look like it's possible to set it on a per-peer/user basis? |
| 23:26.59 | jkroon | or am i missing something? |
| 23:27.23 | citywok | is there any wya to get the meetme confno out of the meetme application? |
| 23:28.41 | jkroon | nm, the asterisk code agrees. |
| 23:28.44 | [TK]D-Fender | citywok: When? Where? |
| 23:29.34 | citywok | i want to be able to track which conference codes get used, be it at the beginning of a conference or at the end. I just dont want to have to log in every 60 seconds and use the AGI/AMI to do meetme list |
| 23:30.11 | citywok | it's not a channel variable, so i can't get it any wya i know how |
| 23:30.14 | [TK]D-Fender | citywok: How are you prompting for the room #? |
| 23:30.19 | citywok | and googling has not been helpful |
| 23:30.33 | citywok | right now it's Answer, MeetMe(|ipscM) |
| 23:30.57 | citywok | i let meetme handle getting the code. i've considered prompting the user myself, and then passing the code in to meetme, but it seems as though there should be a built in way of doing this. |
| 23:31.04 | [TK]D-Fender | City, then prompt for the room YOURSELF and pass it to MeetMe and log it |
| 23:31.30 | [TK]D-Fender | citywok: Well There isn't. You go into MettMe, and you're not coming out. |
| 23:31.45 | citywok | kk, good to know. |
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| 23:54.20 | citywok | thank you :) |
| 23:55.45 | *** join/#asterisk Faiz (n=otakucon@c-69-253-143-177.hsd1.nj.comcast.net) |
| 23:55.48 | Faiz | hi everyone |
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| 23:56.54 | Faiz | i have a newbie question if someone doesn't mind? |
| 23:57.18 | Faiz | Reading the documentation suggests getting the TDM11B card for beginners |
| 23:57.33 | Faiz | since this model is now discontinued, what would be the best card to purchase? |
| 23:58.50 | ManxPower | Faiz: anything with 1 FXS and 1 FXO |
| 23:59.38 | Faiz | does it matter if its a TDM or not? |
| 23:59.47 | Faiz | for beginners, should i opt for the cheap ebay ones? |
