| 00:00.22 | KyleK | Odd, i'm getting this even with just format=wav [Jun 16 17:01:32] ERROR[25934]: utils.c:1126 ast_carefulwrite: write() returned error: Broken pipe |
| 00:00.36 | KyleK | ast_nottoocarefulwrite? ;) |
| 00:01.45 | pfn | KyleK, wav49 |
| 00:01.49 | pfn | ? |
| 00:02.02 | pfn | hmm |
| 00:02.24 | dacs | can someone please take a look and tell me what i missed http://pastebin.ca/1462793 |
| 00:02.47 | dacs | i also did sip reload and dialplan reload |
| 00:02.55 | pfn | dacs, context default does not have 1000 |
| 00:03.14 | pfn | or, in this case, context phone != context phones |
| 00:03.41 | dacs | pfn: can you please explain more |
| 00:03.54 | pfn | you say context=phones, whereas you have [phone] |
| 00:04.27 | drmessano | context=phones |
| 00:04.29 | drmessano | ha beat me to it |
| 00:04.44 | drmessano | fail |
| 00:04.46 | comradeb14ck | hi all--in asterisk 1.6 AGI scripts. if I use the CHANNEL STATUS command and want to specify a paramater, do i put it in quotes or not? eg: CHANNEL STATUS SIP/blah or CHANNEL STATUS "SIP/blah" ? |
| 00:05.08 | comradeb14ck | i cant find that documented anywhere, but it appears that if i dont specify the channel, eg: CHANNEL STATUS\n , it will ONLY work without the quotes |
| 00:05.29 | comradeb14ck | so i just want to make sure (because i cant find docs anywhere) that all commands cannot use quotes |
| 00:05.40 | comradeb14ck | thanks ^^ |
| 00:05.52 | dacs | pfn: can you please explain it to me, i swear everytime i see the word context my head spin |
| 00:06.07 | pfn | dacs, the words need to match |
| 00:06.15 | pfn | if they don't match, you no go |
| 00:06.58 | dacs | pfn: so in sip.conf the context=phones is true |
| 00:07.08 | pfn | dacs, in extensions.conf it's wrong |
| 00:08.08 | dacs | pfn: oh, so i need to remove all what is under internal tab and put it under [phone] |
| 00:11.20 | dacs | pfn: i am sorry but please explain it to me |
| 00:12.24 | *** join/#asterisk layne (n=layne@ool-44c0048f.dyn.optonline.net) |
| 00:12.41 | pfn | contexts define what extensions a user is allowed, if you don't set the right context, then you will not find the right extension, all there is to it |
| 00:12.54 | pfn | you have a phone context in your dialplan, yet you set context=phones in sip.conf |
| 00:13.07 | pfn | if you aren't clued in enough by that statement, repeated more than 3 times now probably, ... |
| 00:14.11 | KyleK | sweet i get caller id |
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| 00:19.50 | dacs | pfn: thank you, that made sense to me, i changed the context=phone in my sip.conf, to context=internal on both my extention |
| 00:21.10 | pfn | sighs |
| 00:21.17 | pfn | you didn't set it to context=phone in your sip.conf |
| 00:21.20 | pfn | you set it to context=phones |
| 00:23.24 | dacs | wtf |
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| 00:25.59 | dacs | i can call from 1000 to 2000, but not from 2000 to 1000 |
| 00:26.58 | dacs | -- Registered SIP '2000' at 192.168.1.104 port 5076 |
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| 00:35.21 | infernix | Qwell: hate to be a drag, but.. chan_mobile + android, any luck yet? still looking for another soul on the planet that tried it |
| 00:36.07 | infernix | apparently nobody else is, nobody in #android at least |
| 00:36.23 | pfn | do you people sit close enough to your asterisk box to make it worthwhile? :p |
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| 00:39.39 | pfn | maybe in addition to installing cat6 drops in my house, I should also install usb drops so I can run a long ass usb cable from my server closet to my rooms with bluetooth dongles |
| 00:40.40 | carrar | I sit 11.4 miles from my asterisk boxes |
| 00:40.56 | dacs | pfn: u can use Cat5 for that |
| 00:41.11 | pfn | carrar, and I suspect you aren't using chan_mobile |
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| 00:42.16 | carrar | You are correct in your suspicions |
| 00:42.37 | pfn | cool concept, just so darn impractical |
| 00:45.46 | dacs | pfn: thank you i changed my context to phone and it works i was able to make test calls between the 2 extension i have |
| 00:45.57 | dacs | time for some more more reading |
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| 01:07.36 | KyleK | pfn: my server is in a closet at the center of my apartment |
| 01:08.25 | pfn | KyleK, that's kind of iffy for bluetooth, unless your apartment is pretty small |
| 01:08.47 | pfn | 30' range isn't much, and walls cut it down pretty quickly |
| 01:08.49 | KyleK | its bigger than some but its not huge |
| 01:09.09 | infernix | my asterisk box is downstairs |
| 01:09.14 | KyleK | yea the width of my place is like 50' |
| 01:09.41 | infernix | i have a linksys sip phone with a headset. having one headset for all incoming calls is my main reason to hook my cell to asterisk with chan_mobile |
| 01:10.10 | infernix | keeps me from having to switch headsets 30 times a day :p |
| 01:10.35 | pfn | can you do chan_mobile with just the headset and without the cellphone? that sounds kinda cool, if it weren't for the range |
| 01:10.45 | infernix | i just set up chan_mobile on my desktop and connect that asterisk to asterisk on my server |
| 01:10.52 | infernix | yes it can do that too |
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| 01:10.56 | pfn | infernix, it'd be nicer if the IP phone supported bluetooth headsets |
| 01:11.00 | infernix | although originating calls is done on asterisk -r |
| 01:11.15 | infernix | pfn: that still wouldn't route my incoming cellphone calls into asterisk, though |
| 01:11.47 | infernix | and i could order a bluetooth headset for my sip phone but i like wires :) |
| 01:12.11 | infernix | plus they're really expensive |
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| 01:14.30 | infernix | i did try the range with some bluetooth usb sticks hooked to the server downstairs |
| 01:14.44 | pfn | how'd that turn out? |
| 01:14.45 | infernix | but it doesn't even pass through one wall |
| 01:14.55 | pfn | and there it goes... heh |
| 01:15.07 | infernix | so unless you wanna do your calls in your server room, forget that :p |
| 01:15.38 | pfn | yeah, my server room is my garage... no go there |
| 01:15.48 | infernix | i did read some posts about people running it on routers which had asterisk and usb ports |
| 01:15.54 | infernix | that should work nicely |
| 01:15.59 | pfn | I need to hack my 7960 so that I don't need a lifter when turning my headset on |
| 01:16.09 | pfn | infernix, that'd be pretty cool if it worked on the sheevaplug |
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| 01:16.56 | infernix | pfn: i don't see why it shouldn't |
| 01:17.11 | infernix | 1.2ghz arm should be fine |
| 01:17.19 | infernix | even for bt audio decoding |
| 01:18.05 | pfn | not bad then |
| 01:18.12 | infernix | that is a nice wall plug btw |
| 01:19.57 | infernix | "Note: At the moment we just have a version for U.S. wall socket; power adaptor may be required to take this US plug to fit for other countries". doh. |
| 01:20.26 | infernix | there goes the 5 watt usage. im so not gonna plug it in a converter |
| 01:21.54 | KyleK | oh is there a wiki doc somewhere that i should note down to use blackberry=yes for a motorola v3 edge? |
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| 01:31.06 | KyleK | okay I edited http://www.voip-info.org/wiki/view/chan_mobile |
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| 02:18.40 | pfn | infernix, I take it that means you're not us based :p |
| 02:21.38 | KyleK | i forgot about the sheevaplug |
| 02:21.47 | KyleK | i keep thinking of getting an alix boar |
| 02:21.50 | KyleK | *board |
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| 02:22.47 | WhIteSidE | Hello all, I'm having a little trouble setting up a TDM400P and DAHDI |
| 02:23.31 | WhIteSidE | Specifically, whenever I run "DAHDI RESTART" It replies "ERROR[2694]: chan_dahdi.c:12031 setup_dahdi: Unable to load zapata.conf |
| 02:24.15 | WhIteSidE | However ztcfg runs fine |
| 02:24.38 | WhIteSidE | Do I need to manually create zapata.conf? I thought that the config did so automatically? |
| 02:24.42 | KyleK | hrm i thought they finished changing over to all dahdi references |
| 02:25.23 | WhIteSidE | I'm running 1.4 svn trunk (200513) |
| 02:26.49 | WhIteSidE | should I try manually creating zapata.conf? |
| 02:27.19 | KyleK | go for it |
| 02:27.49 | KyleK | i personally just use sip and chan_mobile so i can only guess at the dahdi stuffs |
| 02:27.57 | srf21c | ~dahdi |
| 02:27.57 | infobot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi , and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
| 02:31.34 | srf21c | I could use some help troubleshooting a Dial application hangup issue. |
| 02:32.33 | srf21c | For some reason, when I set the dial application timout to 25 seconds or greater, the asterisk will hang-up the incoming call and then immediately re-accept the same call from my ITSP. |
| 02:33.08 | srf21c | So instead of ringing for 25 seconds and going to voicemail, it will ring 25 seconds, appear to stop for a second, then ring another 25 seconds, *then* go to voicemail. |
| 02:33.38 | srf21c | If I set the dial application timeout value in the dialplan context to 24 seconds or less, the problem goes away. |
| 02:33.45 | WhIteSidE | KyleK: and srf21c: Interesting, based on my install, it looks like the flag to use "zap" names was set |
| 02:33.59 | srf21c | and incoming calls go the voicemail as expected after reaching the dial application timeout value. |
| 02:34.04 | WhIteSidE | It must have been part of the configuration, because I don't think I set it |
| 02:34.29 | srf21c | WhIteSidE: glad you found the problem. |
| 02:34.37 | WhIteSidE | Well, I hope so |
| 02:34.58 | WhIteSidE | For what it's worth, I had the exact same problem you're having yesterday, and a restart fixed it |
| 02:34.59 | srf21c | break out the champagne. ;) |
| 02:35.04 | srf21c | really? |
| 02:35.05 | KyleK | srf21c: do you have a sip phone? |
| 02:35.11 | srf21c | I have a suspicion that it's a bug. |
| 02:35.12 | srf21c | yes. |
| 02:35.30 | KyleK | make an extension go through the same logic |
| 02:35.39 | srf21c | ok, good idea. |
| 02:35.47 | WhIteSidE | fcuk, all I changed was dahdichanname = no to dahdicanname=yes and now asterisk won't start |
| 02:35.50 | srf21c | that way I can isolate between SIP and my IAX provider. |
| 02:35.52 | KyleK | like exten => 1234,1,GoTo(.... |
| 02:35.56 | KyleK | yea |
| 02:36.02 | srf21c | ok standby. |
| 02:39.28 | *** part/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 02:40.31 | srf21c | so..... |
| 02:41.00 | srf21c | if I dial from extension to extension, dial application timeouts over 24 seconds to do cause the problem. |
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| 02:42.15 | AlmightyOatmeal | does someone have a cisco.com user account that can grab some ip phone firmware for me? |
| 02:43.25 | WhIteSidE | Hmm, I thought that maybe I had compiled without dahdi, but running menuselect it looks like I did make it with chan_dahdi. Recompiling just to be sure |
| 02:43.52 | srf21c | AlmightyOatmeal: what, you don't have measly few thousand dollars to shell out for a support contract? :-B |
| 02:44.05 | AlmightyOatmeal | srf21c: no, hence why i'm asking ;P |
| 02:44.23 | AlmightyOatmeal | i'm only a poor kid with an ip phone :'( |
| 02:44.31 | srf21c | AlmightyOatmeal: I was just goofin'. One of the reasons I don't like Cisco... |
| 02:44.43 | srf21c | they hold you hostage to expensive contracts just to get software updates. |
| 02:44.50 | AlmightyOatmeal | yeah, negative points for cisco for hiding their sip firmware from me |
| 02:44.58 | KyleK | i still haven't looked into serious IP phones yet, i'm still happily using SPA3102's ;) |
| 02:45.10 | srf21c | although they do make the best ethernet switches from what my net admin buddies tell me. |
| 02:45.22 | AlmightyOatmeal | i was thinking of picking up a couple of cheap sip phones on ebay for <= $50 |
| 02:45.29 | srf21c | werd |
| 02:46.13 | AlmightyOatmeal | >= $50 * |
| 02:46.25 | AlmightyOatmeal | but i need to save up some freaking allowance heh |
| 02:46.41 | KyleK | im in canada so ordering from ebay is always tricky |
| 02:46.53 | AlmightyOatmeal | well yeah, you're canaidan |
| 02:47.38 | AlmightyOatmeal | hides |
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| 02:55.11 | drmessano | any reason why alwaysauthreject=yes would prohibit ALL calls? |
| 02:56.52 | drmessano | I seem to be getting a bearer capability not available when I check debug |
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| 02:58.42 | wwalker | I'm getting this trying to load chan_dahdi. anyone else seen it? google hasn't. Fresh build and install loader.c:400 load_dynamic_module: Error loading module 'chan_dahdi.so': /opt/asterisk-1.4.25.1/lib/asterisk/modules/chan_dahdi.so: undefined symbol: ast_masq_park_call |
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| 03:00.47 | srf21c | wwalker: I'm still on 1.4.21.2 so prolly not much help to you. |
| 03:02.37 | wwalker | we're building from scratch again, but expect similar results. |
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| 03:09.51 | srf21c | KyleK: I'm going to restart asterisk now, see if that resolves the dial application timeout issue. |
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| 03:16.52 | srf21c | nope, restart did not solve the issue. dang. |
| 03:17.00 | srf21c | KyleK: you said you had the exact same issue? |
| 03:17.49 | drmessano | carrar: |
| 03:20.03 | KyleK | nope |
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| 03:34.12 | srf21c | KyleK: sorry, that was whiteside |
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| 03:36.15 | WhIteSidE | Ok, let me run my install process by, see if I'm doing something wrong |
| 03:36.39 | WhIteSidE | Get svn versions of zaptel, libpri and asterisk |
| 03:36.47 | WhIteSidE | (That's /branches/1.4 |
| 03:37.11 | WhIteSidE | configure && make && make install all three |
| 03:37.20 | WhIteSidE | make config zaptel and asterisk |
| 03:37.20 | drmessano | Why not dahdi? |
| 03:37.37 | WhIteSidE | Do I need to install dahdi seperatey? I thought that asterisk pulled that down |
| 03:37.47 | WhIteSidE | If that's my mistake I'm going to hit myself! |
| 03:38.53 | drmessano | Dahdi replaced zaptel.. Zaptel is deadware |
| 03:39.23 | WhIteSidE | ok, do I need to remove zaptel, or can I just go ahead and install dahdi and then asterisk |
| 03:39.38 | drmessano | I would suggest installing Dahdi, which removes Zaptel |
| 03:39.52 | drmessano | and in the future, forget zaptel |
| 03:41.46 | WhIteSidE | Ok, thanks |
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| 03:44.48 | hesco | I've been using a Grandstream HT-486 at home as my office phone now for six or eight months now |
| 03:45.09 | hesco | this is the first time I've taken it on the road. |
| 03:45.30 | hesco | I'm trying to get it to register to my * server back at home. |
| 03:45.39 | hesco | but w/o success so far |
| 03:46.35 | hesco | I updated the SIP server and outbound proxy settings on the device to use the IP of my home network. |
| 03:47.22 | hesco | sip show peers shows the two devices defined in sip.conf |
| 03:47.39 | hesco | but I'm getting no dial tone here |
| 03:48.10 | hesco | Do I need to open some ports on my * server (and its router) at home to make this work? |
| 03:48.17 | hesco | Any clues would be appreciated. |
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| 03:55.09 | srf21c | WhIteSidE: Still having that dial timeout issue. Did you try anything else to fix yours besides server restart? |
| 03:55.31 | WhIteSidE | I set it lower |
| 03:55.41 | WhIteSidE | Sorry, I didn't really find a solution |
| 03:56.26 | WhIteSidE | You might be able to chain two ring groups together, and then goto your secondary action from the second ring group, but this seems really hacky to me |
| 03:57.52 | srf21c | yeah, I have it set lower for now, but would like to get to the bottom of it. |
| 03:58.13 | srf21c | I will try posting to the digium forums, and if that doesn't work, the mailing list. |
| 03:58.28 | WhIteSidE | Well, I'm pretty much a n00b, so I probably can't help that mich |
| 03:58.30 | WhIteSidE | *much |
| 03:59.10 | WhIteSidE | In fact, I'm just recompiling because I didn't realize that DAHDI was preferred to zaptel |
| 04:03.03 | KyleK | why not just start with the mailing list? |
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| 04:08.24 | WhIteSidE | Grrr, ok, I can't seem to reconfigure this, so I plan to do a clean install. What's the thought on 1.4 vs 1.6 |
| 04:08.26 | WhIteSidE | ? |
| 04:09.36 | WhIteSidE | I plan to use this as a production machine (eventually) |
| 04:10.51 | srf21c | KyleK: I'm not a big fan of mailing list.. |
| 04:11.06 | srf21c | prefer forum. |
| 04:13.00 | WhIteSidE | anyone? |
| 04:13.02 | KyleK | WhIteSidE: 1.6 is probably stable enough, what all is going to be hooked up to it? SIP/IAX/dahdi, anything else? |
| 04:13.28 | WhIteSidE | SIP DAHDI (a TDM400 and a TE101) |
| 04:13.57 | WhIteSidE | It's for my personal office, though, so service "calls" involve walking to the machine room |
| 04:14.21 | KyleK | may as well do 1.6 |
| 04:14.25 | WhIteSidE | I think I'll go with 1.6. I don't want to get 1.4 configured just in time to upgrade |
| 04:14.40 | WhIteSidE | But I think I'll install the tarballs instead of svn |
| 04:14.45 | KyleK | 1.4 is around because it takes time to upgrade to 1.6, dialplan changes, etc |
| 04:14.58 | KyleK | you can check out a specific release out of svn also |
| 04:15.16 | WhIteSidE | Should I go with /branches/1.6 or /tags/1.6.x |
| 04:16.18 | WhIteSidE | nm |
| 04:16.40 | KyleK | <PROTECTED> |
| 04:17.52 | WhIteSidE | thanks for the help |
| 04:18.04 | WhIteSidE | I'll probably be back again in a couple of hours! |
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| 05:24.41 | nkohh | kamloops (3 hrs out of vancouver) smells like old shoes |
| 05:25.41 | KyleK | so dont go there in summer? |
| 05:25.47 | nkohh | dont go there ever |
| 05:26.09 | nkohh | its hell |
| 05:26.23 | nkohh | its like deep saskatchewan in the midwest |
| 05:26.26 | nkohh | rednecks |
| 05:26.35 | nkohh | of the worst kind |
| 05:27.11 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 05:30.47 | KyleK | hey can I record a call to Voicemailmain? |
| 05:32.19 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 05:50.58 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 06:09.23 | *** join/#asterisk jaguiar (n=jaguiar@201.114.67.164) |
| 06:11.44 | *** join/#asterisk imcdona (n=imcdona@96.9.161.252) |
| 06:11.46 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 06:12.28 | joako | KyleK: I don't see why not... |
| 06:18.52 | imcdona | can anyone recommend a good SIP aware router for $500 or less? I have always used edgewater stuff in the past. IS there any others that I should ocnsider? |
| 06:20.14 | kaldemar | if you use asterisk, you'll probably be better off without a sip aware device. |
| 06:20.45 | kaldemar | that's not the answer you're looking for, but... |
| 06:20.48 | imcdona | what?? how so?? |
| 06:21.18 | imcdona | your talking using the nat option, and just forwarding ports or what? |
| 06:21.32 | kaldemar | then you may have just another variable in your system. and asterisk can be made to function properly without any awareness in the router. |
| 06:22.04 | kaldemar | everything will work with proper nat configuration in asterisk and port forwarding in the router. |
| 06:22.25 | imcdona | so enabling nat in asterisk and forwarding ports? In your opinion that is the best way to go? What about QOS and MOS etc.... |
| 06:23.57 | kaldemar | you can set TOS and 802.1p priority in asterisk. |
| 06:24.31 | imcdona | kaldemar....that is something I didn't consider. I Am going to ponder that thought |
| 06:26.15 | imcdona | what about WiFI voip phones, can anyone recommend a good one? |
| 06:27.57 | *** join/#asterisk xrmx__ (n=rm@host181-252-dynamic.15-87-r.retail.telecomitalia.it) |
| 06:29.29 | kaldemar | http://www.voip-info.org/wiki/view/VOIP+Phones <-- all wifi phones are probably more or less crap. |
| 06:30.01 | kaldemar | i'd consider DECT for the wireless part. |
| 06:31.39 | imcdona | DECT - is that a standard? So I could for example, mix vendors of DECT phones? and why do you recommend DECT? I want the ability to place a call on hold for example |
| 06:34.33 | kaldemar | DECT is a mature old technology known from POTS. the SIP-DECT consumer phones are usually a small base station per phone and not suitful for bigger installations, but there are some SIP-DECT base stations on the market. whether you can mix vendors, i don't know. probably not. |
| 06:42.23 | vousb | blam |
| 06:43.14 | vousb | imcdona: you mean that work universally with sip? |
| 06:43.49 | vousb | aside from using a laptop, I don't know of a bunch. I think they should be more mainstream but they aren't. |
| 06:45.04 | *** join/#asterisk Jords (n=Jords@202-180-119-54.callplus.net.nz) |
| 06:47.43 | Jords | Hmm.. I'm trying to setup a voice menu, and it seems like the touch tones aren't getting through when i call from a POTS line (i have a SIP provider with a pots number to call) - but when i put the same commands (a read() and then SayDigits() to say them) into my internal context's dialplan, it works... |
| 06:47.47 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 06:47.52 | joako | imcdona: SIP clients on Nokia phone is decent |
| 06:48.09 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 06:48.19 | Jords | from the analog line, nothing happens when i push the keys |
| 06:51.22 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 06:52.31 | KyleK | Jords: so your call goes, handset -> wall -> you dial your DID -> asterisk? |
| 06:53.24 | *** join/#asterisk Heretic (n=BuRn@ZA1-securenode.echelon.co.za) |
| 06:53.28 | Jords | yeah, and I get through to astrisk because i can hear what I tell it to play |
| 06:53.42 | Jords | but if i use Background(), it doesn't stop when i push a key |
| 06:53.42 | KyleK | you'll want to look at how your provider does DTMF |
| 06:54.34 | KyleK | whats the sip provider if i may ask? |
| 06:54.47 | Jords | italk in new zealand |
| 06:54.54 | Jords | pretty small one i guess |
| 06:55.16 | KyleK | yea but someones probably used it |
| 06:55.30 | KyleK | do you have dtmfmode=rfc2833 in your sip.conf? |
| 06:57.00 | Jords | yeah, I do, i used a config from here: http://www.wlug.org.nz/AsteriskItalkSetup |
| 06:58.14 | *** join/#asterisk phurl (n=mdupont@84.22.45.7) |
| 06:58.31 | KyleK | hrm, maybe dump the incoming sip request? |
| 06:58.50 | Jords | how do i do that? |
| 06:58.55 | Jords | pretty new to asterisk :) |
| 07:01.07 | KyleK | sip set debug on |
| 07:01.31 | Jords | in asterisk console? |
| 07:01.38 | KyleK | well that gives you all the sip traffic |
| 07:01.39 | KyleK | yea |
| 07:05.02 | Jords | well, i don't seem to get any messages on the console when i push buttons... but i do get ones when i make the call |
| 07:05.55 | phurl | We are looking for good people to come to our software freedom conference in Prishtina Kosovo on the 29/30 of august. SFK09/SFCK09 |
| 07:06.13 | KyleK | yea, well i'm thinking the sip packets might say dtmf is done a certain way |
| 07:08.32 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 07:16.37 | KyleK | http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode try auto maybe? |
| 07:18.28 | *** join/#asterisk lordmortis (n=lordmort@203-8-160-250.secure.com.au) |
| 07:18.40 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 07:22.54 | Jords | doesn't seem to help... very strange |
| 07:22.56 | kaldemar | rfc2833 is not SIP so you won't see it in sip debug. enable debug in logger.conf's console line and you should see incoming dtmf is there is any. |
| 07:24.26 | kaldemar | try info and inband too. if it there are none in any dtmfmode, the matter is probably out of your hands and you need to contact italk. |
| 07:24.40 | kaldemar | -it |
| 07:25.45 | Jords | i've got debug in the console line and restarted asterisk, i don't see anything in the console when i call my number... doesn't seem right |
| 07:26.53 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 07:27.02 | kaldemar | and you have verbosity in the console? as with command core set verbose 10 |
| 07:27.20 | Jords | oh no havn't will do |
| 07:28.09 | kaldemar | verbosity is 0 by default unless you enable some in the startup or in the console. without verbosity, asterisk doesn't show pretty much anything about what's going on. |
| 07:30.31 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:30.34 | Jords | ok, it's not showing anything when i press the buttons but is saying it's executing the read command. so I'll try those other dtmf options... |
| 07:31.02 | *** join/#asterisk sergee (n=serg@195.94.224.197) |
| 07:31.12 | KyleK | also record a call |
| 07:33.00 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 07:34.46 | Jords | ok, cool, dtmfmode=inband is working :) |
| 07:34.52 | *** join/#asterisk sergee (n=serg@195.94.224.197) |
| 07:34.55 | Jords | thanks |
| 07:52.39 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:c85c:ecd1:f06b:acab) |
| 07:55.31 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
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| 08:00.52 | *** join/#asterisk ludan (n=ludan@unaffiliated/ludan) |
| 08:01.00 | ludan | hi guys |
| 08:01.08 | ludan | I'm trying to use twinkle to connect to my asterisk server |
| 08:01.19 | ludan | from phonerlite or ekiga everything works |
| 08:01.30 | ludan | but from twinkle I get: 502 Bad Gateway, We don't relay to other proxies |
| 08:01.32 | ludan | blue |
| 08:01.40 | ludan | don't know why, any clue? |
| 08:03.27 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:03.45 | KyleK | huh cepstral forums have a bunch of spam, and the software hasn't been updated since june 11th 2008, are these guys still around? |
| 08:05.52 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 08:06.12 | *** join/#asterisk mbranca (n=matteo@2001:1418:130:0:21e:8cff:fe51:5b05) |
| 08:07.52 | kaldemar | ludan: where is the 502 coming from? |
| 08:08.32 | ludan | kaldemar: from twinkle |
| 08:08.47 | ludan | kde sip phone sw |
| 08:09.35 | kaldemar | it generates it? |
| 08:10.05 | ludan | kaldemar: I can login on the asterisk server |
| 08:10.09 | ludan | I can see it from the logs |
| 08:10.23 | kaldemar | you mean register to asterisk? |
| 08:10.25 | ludan | but when I try to connect to the meetme conf (number 8900) |
| 08:10.32 | ludan | yes, register |
| 08:10.41 | ludan | it says that from twinkle |
| 08:10.54 | ludan | if I try from a different sw it just works |
| 08:10.57 | ludan | blue |
| 08:11.42 | kaldemar | don't look at twinkle, look at asterisk. sip show peers to verify the registration and look at the cli when you make the call. |
| 08:11.57 | ludan | ok |
| 08:11.58 | ludan | wait |
| 08:12.03 | ludan | I'll enable debug mode |
| 08:12.09 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 08:12.15 | ectospasm | debug may give TMI at this point |
| 08:12.41 | ludan | <PROTECTED> |
| 08:13.00 | ludan | Name/username Host Dyn Nat ACL Port Status |
| 08:13.00 | ludan | ludan/ludan 82.210.xxx.xxx D N 5642 Unmonitored |
| 08:13.10 | ludan | so it works the registration, doesn't it?! |
| 08:13.35 | kaldemar | yes. now what happens when you make a call. start with verbosity and pastebin the output. |
| 08:13.59 | ludan | how to enable verbose? I forgot |
| 08:14.07 | ectospasm | ludan: core set verbose 3 |
| 08:14.17 | ludan | thanks |
| 08:14.39 | ludan | :( |
| 08:14.42 | ludan | no output at all |
| 08:14.48 | ludan | how come?? |
| 08:15.10 | kaldemar | enable sip debug and call again. if there is no output, then the call is going somewhere else. |
| 08:15.12 | ludan | it doesn't reach the asterisk server at all |
| 08:15.56 | kaldemar | asterisk doesn't generate 502 messages, so either some other proxy/UAS is giving it to twinkle, or twinkle is making it up, which doesn't make much sense. |
| 08:16.12 | ludan | 503 Service not available - No gateways Warning: 392 Noisy feedback tells: pid=9898 req_src_ip=82.210.xxx.xxx req_src_port=5642 in_uri=sip:600@voip.eutelia.it out_uri=sip:600@voip.eutelia.it via_cnt==1 (83.211.xxx.xxx:5060) |
| 08:17.14 | kaldemar | that's from twinkle? you're calling voip.eutelia.it directly. use only the number when calling. |
| 08:17.18 | ludan | yes |
| 08:17.42 | kaldemar | that's a mess in twinkle configuration. |
| 08:19.12 | ludan | Contact: <sip:ludan@192.168.2.102>;expires=3600 |
| 08:19.15 | ludan | here you go |
| 08:19.24 | ludan | from asterisk log |
| 08:19.28 | ludan | maybe it's the NAT |
| 08:19.55 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 08:20.44 | kaldemar | that contact header tells nothing at all. make twinkle send calls to asterisk, not somewhere else. |
| 08:21.20 | ludan | I'm behind a router |
| 08:21.26 | ludan | shall I enable something in twinkle? |
| 08:21.29 | ludan | I don't think so... |
| 08:21.51 | ludan | I don't have a stun server |
| 08:22.30 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 08:23.19 | kaldemar | enable nat=yes and qualify=yes in asterisk for your peer. |
| 08:23.26 | ludan | fuck me |
| 08:23.27 | ludan | wait |
| 08:23.29 | kaldemar | and disable any outbound proxy in twinkle. |
| 08:23.33 | ludan | idiot |
| 08:23.41 | ludan | i didn't select the right user configuration |
| 08:23.45 | ludan | in twinkle |
| 08:23.47 | ludan | blueee |
| 08:23.51 | ludan | what an idiot |
| 08:23.54 | ludan | but |
| 08:23.55 | kaldemar | shame on you. ;) |
| 08:24.10 | ludan | I was fool because I didn't register the other account I have in twinkle |
| 08:24.24 | ludan | so I guessed the only possible account enabled was the asterisk one :D |
| 08:24.30 | ludan | thanks anyway guys |
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| 08:46.48 | Boardy | Hello |
| 08:47.19 | Boardy | Does anybody know everything about MOH? |
| 08:47.42 | Boardy | (Or at least more than me, which should be easily achieved.) |
| 08:48.10 | tzafrir_laptop | As we don't know how much you know, we really can't answer that |
| 08:48.33 | tzafrir_laptop | And mr. Anybody isn't here, which makes things worse |
| 08:48.33 | Boardy | well... I don't know much. |
| 08:48.47 | tzafrir_laptop | But if you actually have a specific question, ask ahead |
| 08:48.48 | KyleK | !ask |
| 08:48.58 | Boardy | then I guess I was looking for anybody else. ;) |
| 08:49.00 | KyleK | Boardy: go into #ubuntu and type !ask ;) |
| 08:49.02 | tzafrir_laptop | ~ask |
| 08:49.02 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 08:49.49 | tzafrir_laptop | notes KyleK here is against his will |
| 08:50.15 | Boardy | Ok... Anyway... I got MOH working on Asterisk (lenny distribution). But when I check my processes I see a couple of mpg123 processes |
| 08:50.34 | tzafrir_laptop | you don't need mpg123 for moh on asterisk |
| 08:50.35 | Boardy | When I restart asterisk, these often become more |
| 08:50.41 | KyleK | tzafrir_laptop: hehe I like my bofh response ;) |
| 08:50.45 | tzafrir_laptop | do you happen to use safe_asterisk? |
| 08:51.12 | Boardy | Not that I know... I just use the regular asterisk package. |
| 08:51.36 | tzafrir_laptop | Boardy, make sure you use: 'mode = files' (which should be the default anyway, IIRC) |
| 08:51.45 | Boardy | In my config I use mode=quietmp3 |
| 08:52.43 | Boardy | Ok... I'll test with that. |
| 08:52.47 | Boardy | thanks for the help. |
| 08:54.35 | tzafrir_laptop | transcoding mp3 files is a waste of CPU cycles |
| 08:54.49 | tzafrir_laptop | convert them off-lines to wavs |
| 08:55.07 | tzafrir_laptop | typically it won't even waste you disk space |
| 08:55.31 | Boardy | what settings for the wavs? |
| 08:56.12 | tzafrir_laptop | just play them. But they have to be mono, 8kHz, 16 bits per sample |
| 08:56.36 | *** join/#asterisk gunter (n=user@87.127.97.39) |
| 08:56.43 | Boardy | (That was exactly the problem, because it seems unable to find the files...) |
| 09:04.43 | *** join/#asterisk sergee (n=serg@195.94.224.197) |
| 09:09.32 | Boardy | Perfect. It works. Thanks a lot. (And another question I had is solved as well) |
| 09:10.31 | *** join/#asterisk imcdona (n=imcdona@96.9.161.252) |
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| 09:43.54 | *** join/#asterisk chrisdus (n=chrisdus@217.112.144.109) |
| 09:45.09 | chrisdus | hi guys... have a question on how to attach my 2 ISDN lines to my B410P card.. Should I keep the ST boxes in between, or should I take them out? And should I set the card to TE, or keep it at NT mode? |
| 09:46.50 | tzafrir_laptop | if you want to connect lines, you need to use TE mode with your card |
| 09:50.10 | chrisdus | great thanks... and do you know if I should take out the ST devices or keep them on the line? (If yes, I assume I need only one line per span from the ST to one of the interfaces on the card..?) Hope you don't mind me asking, I am new to this! :D |
| 09:50.23 | angryuser | chrisdus, keep boxes and set to TE |
| 09:50.33 | chrisdus | fantastic angryuser! |
| 09:52.30 | angryuser | chrisdus, i assume that "ST boxes" are your telco installed boxes ? If not explain what it is |
| 09:55.42 | chrisdus | Yes, ST boxes have been installed by the telco.. it receives one cable (U-Line/ span) and has two ports on the other side with "ST" as description next to those two ports.. |
| 09:56.03 | chrisdus | .. if ST box is the correct word..! (..?) |
| 09:56.19 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 09:57.37 | kaldemar | S/T would be better. |
| 10:01.14 | chrisdus | thanks kaldemar.. |
| 10:01.41 | chrisdus | I am going to sert it up like you guys described... |
| 10:10.25 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 10:12.54 | angryuser | chrisdus, ok, so what is said stands, you can use a simple cat 5 cables to connect your ports. (not a cross over) |
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| 10:44.55 | clive- | powerkill hi |
| 10:45.06 | PoWeRKiLL | hi |
| 10:52.07 | chrisdus | thanks for the tip angryuser.. say, would you happen to know the australian line encoding settings I need to set in the system.conf? |
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| 11:03.42 | angryuser | chrisdus, no |
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| 11:06.58 | dovid | morning ev1 |
| 11:07.20 | clive- | hey dovy |
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| 11:24.16 | Faxe78 | Hi, I want to use "originate" in the cli to start an outgoing Fax (using "SendFax"). How can I do that? Do I need a dummy device handling this call or can I start it from anywhere in the dialplan? Any hint in the right direction is appreciated. |
| 11:24.58 | Faxe78 | Asterisk version is 1.6.0.9 |
| 11:26.45 | *** join/#asterisk cjk (n=cjk@vodsl-9494.vo.lu) |
| 11:32.23 | *** join/#asterisk Blackvel (n=blackvel@88.65.105.77) |
| 11:32.27 | Blackvel | hi all |
| 11:33.43 | cjk | hi, when i redirect a call with an old isdn phone to my mobile i see the original callerid and an information that it is a forwarded call. how can i achieve this with asterisk |
| 11:34.40 | Blackvel | what is the best way to do inside a context D when steps should be/shouldn't be conditional executed (e.g Playback) depending on context callers (context goto). ctx A->C->D, ctx B->C->D? |
| 11:35.09 | Blackvel | the playback in context D should be only executed when comming from context A, not context B |
| 11:35.27 | Blackvel | A and B share a common context C |
| 11:35.36 | Blackvel | setting a variable? |
| 11:39.53 | *** part/#asterisk srf21c (n=seth@ip98-165-60-42.ph.ph.cox.net) |
| 11:40.26 | leifmadsen | Blackvel: just set a variable |
| 11:41.01 | Blackvel | cjk: had the same problem |
| 11:41.01 | leifmadsen | Exec(${IF($[${EXISTS(${MY_VARIABLE})}]?Playback(tt-weasels):NoOp(Variable not set))}) |
| 11:41.57 | leifmadsen | Faxe78: use the Originate() dialplan application |
| 11:42.05 | Blackvel | leifmadsen: ok, just was wondering if that this is the "usual" way (maybe there was an option like asking what the starting context was) :) |
| 11:42.31 | leifmadsen | Blackvel: huh? I don't understand what you mean the "usual" way |
| 11:42.34 | Blackvel | cjk: I didn't find a solution for the forward information |
| 11:42.47 | Faxe78 | leifmadsen: thanks, i found "originate", but I cannot find out how to use it... |
| 11:43.09 | leifmadsen | core show application originate ? |
| 11:43.11 | Blackvel | cjk: what you really need, when your asterisk calls your mobile phone directly is clip no screening |
| 11:43.36 | Blackvel | cjk: not all pstn/isdn equipment is allowed to do that |
| 11:44.20 | Blackvel | cjk: you can check a voip provider if he supports clip no screening. sipgate/germany allows to dynamically set the originating number in callerid(name) with a special international format |
| 11:44.27 | cjk | Blackvel, i think its just not called redirection but transfer and there is an app transfer |
| 11:44.32 | cjk | but it has technical limitations |
| 11:44.39 | Blackvel | cjk: this is how I forward my calls to my mobile phone |
| 11:44.40 | cjk | and for the callerid i can set whatever i want |
| 11:44.41 | cjk | :) |
| 11:44.54 | leifmadsen | what kind of phone are you using? |
| 11:45.52 | Blackvel | cjk: isdn anlageanschluss? |
| 11:46.02 | Blackvel | anlagenanschluss |
| 11:46.37 | Faxe78 | leifmadsen: no, you get the help for originate by typing "originate" in the cli. I read this anyway. I got this: |
| 11:46.50 | Faxe78 | originate capi/g1/012345 extension in2mail@custom-fax-in |
| 11:46.58 | cjk | Blackvel, no, SS7 interconnection |
| 11:46.59 | leifmadsen | Faxe78: no no... I'm saying use the Originate() dialplan application -- not the CLI command |
| 11:47.20 | leifmadsen | Faxe78: if you want to trigger an Originate from the dialplan in 1.6, you use the dialplan application, not the CLI command |
| 11:48.04 | Faxe78 | core show application has no Originate... |
| 11:48.12 | Faxe78 | but i want to call it from the CLI |
| 11:49.09 | Faxe78 | i just managed to dial out with the command above - how can I give parameters (like the TIF-File to fax)? |
| 11:51.01 | leifmadsen | Faxe78: hmmmm.... pretty sure app_originate is in 1.6.0 -- maybe it's not selected in menuselect. Additionally, you could try: help originate. That may give you more information. I don't think you can pass a variable with that command -- not really what it's meant for. What you *really* want is the dialplan Originate() command |
| 11:51.33 | Faxe78 | ok, i'll have a quick look in menuconfig |
| 11:54.45 | Faxe78 | no, no app_originate to select |
| 11:55.48 | *** join/#asterisk doolittlework (n=f@196.211.34.2) |
| 11:55.57 | doolittlework | sip show peers |
| 11:56.05 | doolittlework | lol wring terminal |
| 11:56.10 | doolittlework | wrong |
| 11:56.18 | doolittlework | hi there |
| 11:56.23 | doolittlework | anyone active? |
| 11:57.33 | kaldemar | leifmadsen: no app originate in 1.6.0 or 1.6.1 |
| 11:58.31 | Faxe78 | anyway: how would app_originate help me on sending a fax? i don't get the clue... |
| 11:59.08 | *** join/#asterisk bsilberman (n=bsilberm@65.213.221.252) |
| 11:59.28 | doolittlework | is it possible to break into a active call and talk to both parties? |
| 12:00.02 | *** join/#asterisk jeffik (n=jeffik@69-196-165-181.dsl.teksavvy.com) |
| 12:00.22 | leifmadsen | Faxe78: you are trying to originate from the dialplan, are you not? from your first comment/question it looked like you are trying to execute this from the dialplan, not always from the CLI |
| 12:00.30 | leifmadsen | doolittlework: ChanSpy() |
| 12:01.05 | Faxe78 | currently I'm trying from the cli - later it should be a python app. sorry for confusion. |
| 12:01.06 | leifmadsen | doolittlework: with certain versions of 1.6.x (I forget which -- start with 1.6.0) you can break in like that I'm pretty sure |
| 12:01.22 | leifmadsen | Faxe78: then you should probably be doing this from the Asterisk Manager Interface |
| 12:01.54 | leifmadsen | Faxe78: alternatively (non-elegant): asterisk -rx "originate ... " |
| 12:02.11 | Faxe78 | yes, I tried this too - but I'm not moving forward... |
| 12:02.14 | leifmadsen | kaldemar: really.... I thought that was a feature that went in pretty early |
| 12:02.16 | doolittlework | @leifmadsen: if i use Chanspy() i can only spy on the sip channel i can hear all the parties but only one of them can hear me |
| 12:02.38 | leifmadsen | doolittlework: I think there are flags in newer versions of asterisk to let you speak to both parties |
| 12:02.46 | doolittlework | @liefmadsen is there a way to talk to both parties? |
| 12:02.53 | leifmadsen | see above |
| 12:02.59 | doolittlework | k will check it out |
| 12:03.01 | doolittlework | thx |
| 12:03.43 | kaldemar | leifmadsen: it's in 1.6.2 branch |
| 12:05.14 | leifmadsen | kaldemar: ya, I still haven't memorized where each of the new applications first got introduced |
| 12:05.18 | leifmadsen | thanks |
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| 12:06.40 | Faxe78 | leifmadsen: Just got an Idea how it could work: connect to the asterisk manager interface - use SetVar to set Faxfile, ... - use Originate to initialize the call. Correct path? |
| 12:06.41 | kaldemar | no problemo. it's was in the reviewboard last december so it's about time it's in a branch. :) |
| 12:07.49 | leifmadsen | Faxe78: sounds right |
| 12:07.54 | Faxe78 | Thanks, you saved me hours of research ... |
| 12:08.24 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 12:10.17 | Blackvel | cjk: back....ss7? special equipment / telco provider? |
| 12:10.20 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:10.59 | cjk | Blackvel, yes, but i try to figure out how to do it on bri and pri |
| 12:11.13 | Blackvel | cjk: if you find a way to display "forwarding" on your mobile phone, let me know :) |
| 12:11.45 | cjk | Blackvel, whats your email? |
| 12:12.17 | doolittlework | @liefmadsen:i am running asterisk 1.4.24 is there a way to just update the chanspy application without updating to * 1.6? |
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| 12:20.34 | Blackvel | is there any way to bundle 3 channel techs in DIAL command to one logical phone (pickup wins as normal, BUSY win's too)? the standard behavior seems to be, that BUSY on one phone will let both phones ring |
| 12:21.51 | Blackvel | ChanIsAvail does not work with BUSY? do I really have to upgrade to 1.6 to go for devstate? |
| 12:22.36 | Blackvel | is was wondering if I would use GROUP() in some way to handle this situation |
| 12:23.04 | doolittlework | i am running asterisk 1.4.24 is there a way to just update the chanspy application without updating to * 1.6? |
| 12:24.57 | leifmadsen | Blackvel: you could dial several Local channels, and then have each of those Local channels dial the SIP endpoint, then when one of the Dial()'s (inside Local) came back as BUSY (${DIALSTATUS}) then you could do an Answer() and then that would stop the other phones from ringing, and you could continue on with dialplan logic inside the Local channel that answered |
| 12:25.27 | leifmadsen | doolittlework: if you can backport the application, potentially -- but there are quite a few differences (read, a LOT) in 1.6 verses 1.4 |
| 12:26.05 | leifmadsen | Dial(Local/jimmy@busydetect&Local/johnny@busydetect&Local/sammy@busydetect,30) |
| 12:27.36 | Blackvel | i even use dial with local syntax (delayed ringning on 3rd phone (snom m3) |
| 12:28.09 | Blackvel | answer stops ringing |
| 12:28.12 | Blackvel | ? |
| 12:28.12 | Blackvel | interesting |
| 12:28.39 | Blackvel | any other way? Answer() is one of the first things my IVR does |
| 12:28.57 | Blackvel | so the channel is answered already when the phone interconnection is choosen |
| 12:29.35 | leifmadsen | Blackvel: it'll still work -- you're answering the Local channel |
| 12:29.54 | leifmadsen | you're answering a different channel |
| 12:29.59 | Blackvel | ahhhh |
| 12:30.17 | leifmadsen | ---incoming---> Answer() ---> Asterisk ---> Local ----> Answer() |
| 12:30.20 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 12:30.21 | Blackvel | now I understand |
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| 12:44.18 | talirk81 | anyone know why from agi "set callerid 0000000032" would be failing to set the callerid number? |
| 12:45.00 | [TK]D-Fender | talirk81: Well there is the thought that that variable was deprecated in 1.2 and removed in 1.4 entirely... |
| 12:45.32 | talirk81 | how should i set it in 1.4 from a agi script then? |
| 12:46.01 | talirk81 | i tries SET VARIABLE callerid(num)=0000000032 and SET VARIABLE callerid(num) 0000000032 |
| 12:46.05 | talirk81 | neither worked |
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| 12:55.31 | michael-i | I'm cross-compiling asterisk with uclibc for x86 and the "vfork/fork works" test fails to recognize fork() as working. I want to patch autoconf/ast_func_fork.m4 but it seems its contents are duplicated in menuselect/acinclude.m4. |
| 12:55.55 | michael-i | Should I patch both? Or...how/why exactly is that function duplicated |
| 12:58.50 | [TK]D-Fender | [08:44]<talirk81>anyone know why from agi "set callerid 0000000032" would be failing to set the callerid number? <- While it looks like a variable, this appears to be valid syntax. Only difference is was the samples show this being in all uppercase |
| 12:59.28 | [TK]D-Fender | talirk81: Not sure if this matters, but I might like to see a dump of the value before and after to see how you are testing for it |
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| 13:02.12 | leifmadsen | michael-i: I would suggest you ask the guys in #astlinux as very few (no one) is going to be able to answer your question |
| 13:02.44 | talirk81 | ok ill have to code in a GET VARIABLE to my agi ill let you know when i have a dump of before and after and the command. |
| 13:02.50 | *** join/#asterisk xipi (n=oliver@91-65-13-113-dynip.superkabel.de) |
| 13:03.22 | michael-i | leifmadsen: ok, I'll check there...thanks |
| 13:11.00 | xipi | hi. what is up with "username s"? |
| 13:11.09 | *** join/#asterisk sergee (n=serg@195.94.224.197) |
| 13:11.19 | leifmadsen | xipi: see answer in #asterisk-gui |
| 13:11.27 | leifmadsen | (cross-posting... tsk tsk) |
| 13:12.45 | Blackvel | anyone with snom phones and patton media gateways? |
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| 13:12.59 | tzafrir_laptop | michael-i, menuselect is only used in the build system |
| 13:13.01 | Blackvel | most of the time I have a little delay (no interruption possible) |
| 13:13.37 | tzafrir_laptop | michael-i, and if menuselect gives you real troubles, just use dummy-select |
| 13:13.45 | xipi | leifmadsen: sorry. i will leave this channel. |
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| 13:16.49 | michael-i | tzafrir_laptop: ah, ok...trying a new patch |
| 13:16.59 | tzafrir_laptop | http://git.tzafrir.org.il/?p=asterisk-tools.git;a=tree;f=menuselect |
| 13:19.42 | timeshell_atwork | Happy Threesday! |
| 13:21.42 | eppigy | TRABAJO |
| 13:23.01 | tzafrir_laptop | Wotan's day |
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| 13:25.06 | leifmadsen | jimmy snookah! |
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| 13:49.25 | talirk81 | [TK]D-Fender, http://pastebin.ca/1463412 is the output php functions and dial plan when the callerid isnt changing |
| 13:49.52 | talirk81 | btw i did change it to use uppercase (no change) |
| 13:50.31 | *** join/#asterisk RichiH (i=richih@freenode/staff/richih) |
| 13:50.33 | RichiH | Received VNAK: resending outstanding frames |
| 13:50.39 | RichiH | any idea what that might be? |
| 13:50.52 | michael-i | hi russellb, just as an update to yesterday's struggles, this patch (http://pastebin.ca/1463414) still cannot convince asterisk that I have a fork(). Patching asterisk.c does result in asterisk forking away except then it fails to load any modules for some reason |
| 13:50.54 | michael-i | fun stuff! |
| 13:50.54 | RichiH | google does not seem to be able to decide |
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| 13:53.42 | RichiH | hmm, another question |
| 13:53.54 | RichiH | i have two asterisks |
| 13:54.06 | RichiH | the slave has a rather peculiar ping towards the master |
| 13:54.34 | [TK]D-Fender | talirk81: AGI Rx << SET CALLERID <-- your actual set doesn't look like it passes the number |
| 13:54.53 | RichiH | either ~0.4 ms or ~1.5 ms. not much derivation from those two values. from what i just found, this might cause the VNACKs |
| 13:55.11 | RichiH | and, that is the real question, would this be able to cause: |
| 13:55.18 | RichiH | 1) hacked off sentences, etc |
| 13:55.27 | RichiH | 2) one side not hearing the other at all |
| 13:55.35 | [TK]D-Fender | talirk81: self::AGIEcho("SET CALLERID {$sNumber}"); |
| 13:55.36 | [TK]D-Fender | fwrite(STDOUT, "SET CALLERID {$sVariableName}\n"); |
| 13:55.38 | RichiH | any help appreciated |
| 13:55.44 | [TK]D-Fender | talirk81: Why aren't you using the same var there? |
| 13:55.54 | talirk81 | [TK]D-Fender, i justed noticed that |
| 13:56.19 | talirk81 | and that was it we i refactored for the tests i created that class new and forgot to change the var name |
| 14:00.10 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
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| 14:05.53 | tzafrir_laptop | michael-i, how does astlinux work? |
| 14:06.02 | tzafrir_laptop | IIRC they build with uclinux |
| 14:06.04 | kaldemar | RichiH: http://www.rfc-editor.org/authors/rfc5456.txt <---- 6.3.9 VNAK: Voice Negative Acknowledgement Message |
| 14:07.13 | michael-i | tzafrir_laptop: astlinux uses buildroot + uclibc + busybox. I'm using my own forked version of T2 to build things |
| 14:08.24 | tzafrir_laptop | ok. but specifically, how do they manage to build asterisk |
| 14:08.25 | tzafrir_laptop | ? |
| 14:13.56 | RichiH | kaldemar: aye, i found that (thanks, though!).. so at least the situation where one side does not hear the other might be caused by this, correct? |
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| 14:18.50 | nextime | Hello all. Is there any way to get an "external" authentication for sip peers? or even to add/delete sip peers dinamically from AMI? |
| 14:18.58 | kaldemar | RichiH: might be possible |
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| 14:20.00 | Blackvel | callgroup/pickup group features.conf *8 only works for calls in RINGING state? |
| 14:20.25 | Blackvel | I have a sip peer patton. |
| 14:20.56 | Blackvel | could I take over the caller from IVR to my phone? probably I would have to use other stuff like Steal & Co. |
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| 14:30.31 | RichiH | kaldemar: k |
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| 14:32.22 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 14:40.57 | Blackvel | if the busy handling is the same for exten 100-103 (dailstatus, Answer(), Playtones(busy), Busy(), Hangup(), how can I share that logic for all 4 extensions? macro? |
| 14:41.28 | Blackvel | i have 4 individual dial statemens for all 4 Local extensions |
| 14:42.12 | [TK]D-Fender | Blackvel: yes |
| 14:43.29 | michael-i | tzafrir_laptop: got my patch right to force fork() support. It does fork into the background but again does not load the modules. I'll have a chat with the astlinux guys tomorrow. |
| 14:43.30 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:45.11 | tzafrir_laptop | michael-i, they tend to reside in #astlinux |
| 14:45.51 | tzafrir_laptop | alternatively, you can build asterisk with all the modules already compiled in |
| 14:46.05 | tzafrir_laptop | linked in, rather |
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| 14:47.58 | michael-i | tzafrir_laptop: I'm trying to keep them out of memory until needed. Too bad I'm being VERY successful so far with that ;) |
| 14:50.33 | Blackvel | in a local channel, what is the best way to handle a dial? Hangup (default) for all cases except Busy? Busy-status would be checked separately and handled in macro (GotoIF(DIALSTATUS=BUSY)?lbl_busy, _..,n(lbl_busy),Macro(busy_handling) |
| 14:50.41 | *** join/#asterisk moy (n=moy@74.12.123.90) |
| 14:51.04 | seanbright | assuming i have digium licenses for 192 channels, and a box with 8 1.6GHz cores that only does transcoding, will that be sufficient for g729 calls? |
| 14:51.32 | Blackvel | i mean I do not have to check all states like unvail/noanswer and manually go to Hangup? |
| 14:52.23 | ariel_ | Morning |
| 14:54.45 | [TK]D-Fender | Blackvel: You can do whatever you feel liek. Its your dialplan. |
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| 15:02.25 | phurl | We are looking for good people to come to our software freedom conference in Prishtina Kosovo on the 29/30 of august. SFK09/SFCK09 |
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| 15:16.27 | rue_mohr | ok, if everyone knows so much about these polycom gains, tell me which one I change to make hte handset and speakerphone louder... |
| 15:16.43 | rue_mohr | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg214182.html |
| 15:16.57 | rue_mohr | cause I'm not sure which ones I should be and shouldn't be adjusting |
| 15:18.35 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 15:18.59 | rue_mohr | and I dont know which is recieve and which is transmitt |
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| 15:24.08 | bmoraca | there's a volume button on the phone... |
| 15:24.15 | *** join/#asterisk ingenius (n=alektro@37-98-231-201.fibertel.com.ar) |
| 15:24.30 | bmoraca | push the one with a + next to it while you're in a call and the speaker and handset will get louder |
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| 15:26.13 | Nockian | can someone recommend an outbound sip provider, now that nufone is gone? |
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| 15:28.19 | ramindia | iam getting this error "res_agi.c:2238 deadagi_exec: Running DeadAGI on a live channel will cause problems, please use AGI" on debian |
| 15:28.34 | ramindia | iam googled could not find the solution |
| 15:28.41 | seanbright | ... |
| 15:28.55 | seanbright | change your call from DeadAGI(whatever) to AGI(whatever) |
| 15:28.58 | seanbright | problem solved. |
| 15:29.07 | bmoraca | shhhh |
| 15:29.09 | bmoraca | stop talking sense |
| 15:29.11 | Joel | when a call is parked is there a way to tell if the parkee hungup vs timeout was hit? |
| 15:29.41 | ramindia | yes i have changed but still i get AGI not found |
| 15:30.03 | seanbright | agi not found? |
| 15:30.07 | seanbright | what does that mean? |
| 15:30.12 | ramindia | res_agi.c:2238 deadagi_exec: Running DeadAGI on a live channel will cause problems, please use AGI |
| 15:30.22 | ramindia | Fatal error: Class 'AGI' not found in /usr/share/asterisk/agi-bin/a2billing.php on line 47 |
| 15:30.31 | ramindia | 510 Invalid or unknown command |
| 15:30.38 | seanbright | wow |
| 15:30.46 | spck | anyone know why my sip phones keep registering and saving their useragent |
| 15:30.48 | bmoraca | that means you didn't install a2billing correctly |
| 15:31.02 | Joel | ramindia that's a problem with the php file |
| 15:31.05 | Joel | ramindia not asterisk |
| 15:31.07 | [TK]D-Fender | Sure looks like a missing PHP-AGI |
| 15:32.13 | ramindia | php-agi ? where can i look that packages |
| 15:32.25 | bmoraca | facepalms |
| 15:32.47 | ramindia | i dont see common/lib is nothing there |
| 15:33.22 | ramindia | ls /usr/share/asterisk/agi-bin/lib -al |
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| 15:33.33 | ramindia | <PROTECTED> |
| 15:33.46 | ramindia | but i dont see any library files there |
| 15:34.13 | [TK]D-Fender | ramindia: This is a 3rd party PHP library. Go look where you should have it. This is not a part of * itself |
| 15:34.44 | ramindia | ok thanks |
| 15:39.11 | spck | why do my phones keep saving their useragent, but its not actually saving it in mysql realtime? |
| 15:39.34 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 15:40.22 | [TK]D-Fender | spck: What would that have to do with realtime? |
| 15:40.24 | *** join/#asterisk Aiatek (n=Asterisk@75.112.88.200.m.sta.codetel.net.do) |
| 15:40.43 | [TK]D-Fender | spck: that is a run-time thing, not a peer definition attribute |
| 15:42.00 | *** join/#asterisk hi365 (n=hi365@94.159.177.112) |
| 15:43.50 | spck | it says "Saved useragent "PolycomSoundPointIP-SPIP_330-UA/3.1.3.0439" for peer 7024 |
| 15:43.50 | spck | " |
| 15:43.50 | spck | in the log |
| 15:44.27 | spck | i thought this was typically resaved to the sip.conf file, but in lieu of having that in this case, it should save it in mysql |
| 15:44.41 | spck | specifically the sip_buddies table |
| 15:44.58 | spck | is that not what its supposed to be doing? |
| 15:45.08 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:48.10 | [TK]D-Fender | spck: No. |
| 15:48.22 | [TK]D-Fender | SpSince when does * modify your conf files? |
| 15:48.58 | spck | i'm not sure what to ask |
| 15:49.10 | spck | it keep resaving/reregistering these phones |
| 15:49.24 | [TK]D-Fender | spck: Quick answer is it doesn't. Tracking of what device has last registered to * is a run-time AstDB thing. |
| 15:49.31 | [TK]D-Fender | spck: Not a "configuration" thing |
| 15:49.51 | [TK]D-Fender | spck: And it keeps doing this because your phone keeps reregistering |
| 15:50.06 | spck | any idea why its not putting the Noop in the regcontext? |
| 15:50.51 | [TK]D-Fender | spck: Since I'm not seeing your dump of your dialplan from * CLI nor your configuration, I have no idea. |
| 15:51.00 | Blackvel | is there any way to reduce dialplan lines for extensions 100-104 and busy handling? http://pastebin.com/m42aaffff |
| 15:51.23 | [TK]D-Fender | Blackvel: yes, go make a Macro for this |
| 15:51.25 | spck | i can get that, do you want to see sip.conf? |
| 15:51.43 | Blackvel | done that |
| 15:51.46 | [TK]D-Fender | spck: Thats aprt of what I jsut asked for |
| 15:52.05 | spck | right but i don't know what conf file you want to see |
| 15:52.49 | [TK]D-Fender | Blackvel: You should be doing ALL of the processing in your macro, not just the BUSY part |
| 15:53.04 | Blackvel | but i am not sure if I can reduce that damn lines down to that: exten => 10X,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?macro-busy_handling,s,1) for exten 100, 101, 102, 103 |
| 15:53.05 | [TK]D-Fender | spck: And I just said YES to SIP.CONF |
| 15:54.04 | Blackvel | [TK]D-Fender so the idea would be to move Dial part (different for 100-103 extensions) for the local channel into the macro as well as? |
| 15:54.20 | Joel | Jun 17 10:46:49 WARNING[14400]: app.c:1232 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/8000/Old': File exists --- Would love thoughts on wtf... |
| 15:54.28 | [TK]D-Fender | Blackvel: Even your DIAL should be in your amcro. you should not jsut to it conditionally, and Youa ren't eevn using your macro as a macro, you're just trying to GOTO it like as if it were a dumb context |
| 15:54.44 | Blackvel | would I have to pass the sip peer (snom370/snomm3) to the macro as argument? |
| 15:54.56 | [TK]D-Fender | Blackvel: thats the idea... |
| 15:55.14 | spck | http://pastebin.com/m7bd0c4ee |
| 15:55.28 | Joel | ahh damnit |
| 15:55.30 | Joel | chillin .lock file |
| 15:55.34 | Blackvel | i know that I can call a macro with Macro(name....) |
| 15:55.54 | Blackvel | I had no clue how to combine the macro (busy part) with the gotoIf(BUSY) |
| 15:56.03 | Blackvel | you can hit me.... |
| 15:56.08 | Blackvel | when is a macro a real macro? |
| 15:56.21 | spck | when you can reuse it for something else |
| 15:56.47 | spck | or it saves you time typing |
| 15:56.48 | [TK]D-Fender | Blackvel: exten => 123,1,Macro(dosomething,parm1,parm2,parm3) |
| 15:57.01 | [TK]D-Fender | Blackvel: You need to go read your dialplan basics again |
| 15:57.24 | spck | http://www.google.com/url?sa=t&source=web&ct=res&cd=1&url=http%3A%2F%2Fdownloads.oreilly.com%2Fbooks%2F9780596510480.pdf&ei=XRI5SoeSG8mptgen-cjSDA&rct=j&q=asterisk+telephony+pdf&usg=AFQjCNHH-gNcCSeBwE6DwUNG76JrV64qaw&sig2=Pk_eEn62QcqqPzZt6IYLCQ |
| 15:57.31 | spck | damn google and their links |
| 15:57.56 | spck | http://www.asteriskdocs.org/ |
| 16:00.15 | [TK]D-Fender | ~book |
| 16:00.15 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:00.19 | [TK]D-Fender | Far easier |
| 16:00.31 | spck | heh |
| 16:00.36 | spck | sorry don't know the bot |
| 16:00.49 | spck | its odd i get messages like this: |
| 16:00.49 | spck | <PROTECTED> |
| 16:00.59 | spck | but then it doesn't put anything in sipreg |
| 16:02.14 | comradeb14ck | hi all |
| 16:02.17 | Joel | when a call is parked, and the parkee hangs up while it's in the ring back state, it seems like it's hitting the timeout extension, is this correct? I feel like I should be using a different special extension |
| 16:02.37 | Nugget | "parkee" is a funny word. |
| 16:02.47 | [TK]D-Fender | Nugget: PARKAY! |
| 16:02.52 | [TK]D-Fender | ....butter |
| 16:02.59 | Nugget | fights for his right to parkaaaaaaaaaaay |
| 16:03.00 | [TK]D-Fender | spck: Sure it does.... |
| 16:03.20 | Joel | parkee vs parker :D |
| 16:03.22 | spck | if i do dialplan show sipreg, its empty |
| 16:03.27 | Nugget | I can't believe it's not trixbox! |
| 16:03.41 | [TK]D-Fender | spck: And you still aren't showing me anything |
| 16:04.36 | rue_mohr | bmoraca, the volume control does not have enough range to bring the volume up to the point where its reasonable |
| 16:05.02 | rue_mohr | for contrast I had to change the analog gain on the polycom from 0 to 8 to get to reasonable |
| 16:05.30 | spck | please tell me what you would like to see |
| 16:05.36 | spck | because i'm at a loss here |
| 16:06.17 | rue_mohr | I'v "properly calibrated the dahdi rx levels such that 1mw obtains about 14844 when using -vv with dahdi_monitor" |
| 16:06.38 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 16:07.00 | *** join/#asterisk s14ck (n=s14ck@190.76.82.176) |
| 16:07.39 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 16:08.08 | rue_mohr | but with over 10db on the dahdi tx I cant get the rx level to 14844 when I have a call looped through the co |
| 16:08.30 | rue_mohr | so I'm leaving it at -5 where I had it, and were still getting unaceptable echo with some calls |
| 16:09.20 | rue_mohr | I have to go, back later |
| 16:09.30 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 16:13.46 | *** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net) |
| 16:14.14 | DavidR2008 | is there any easy way to identify all US holidays in the dialplan? |
| 16:15.18 | *** join/#asterisk Blackvel (n=blackvel@dslb-088-065-105-077.pools.arcor-ip.net) |
| 16:15.27 | Blackvel | laptop battery empty... |
| 16:21.45 | *** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 16:22.23 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 16:22.35 | [TK]D-Fender | DavidR2008: Identify how? Do what with it? Where? |
| 16:23.24 | *** part/#asterisk icyValk77 (n=icyValk7@host217-43-31-121.range217-43.btcentralplus.com) |
| 16:23.26 | [TK]D-Fender | rue_afk: What are your DAHDI gains at now? |
| 16:24.53 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 16:25.09 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:25.39 | DavidR2008 | [TK]D-Fender: I wanted to be able to modify an IVR menu based on holidays, I found a sample that should do what I need on voip-info.org, it just seemed that holidays would be such a common thing that it might be built-in. (i.e. ${HOLIDAY} = 1 if it's a holiday or something) |
| 16:26.55 | [TK]D-Fender | DavidR2008: It isn't Hapy coding |
| 16:27.04 | [TK]D-Fender | DavidR2008: It isn't. Happy coding |
| 16:27.07 | DavidR2008 | :-D |
| 16:30.56 | bmoraca | DavidR2008: how many holidays are on the same day every year? how many holidays are all businesses closed for worldwide? there's no way to do that, except localized in each individual dialplan |
| 16:31.49 | DavidR2008 | yea, I know, but "hope spring eternal" ... ;-) |
| 16:32.02 | DavidR2008 | *"hope springs eternal" |
| 16:32.08 | [TK]D-Fender | waits for it to pool, then holds DavidR2008's head under |
| 16:32.28 | DavidR2008 | drowns |
| 16:32.51 | *** join/#asterisk pta200 (n=paolo@69.38.173.162) |
| 16:33.09 | Aiatek | he can start finding all the holydays in the world and get into that variable, later you can choose your default zone to load yours proper holydays |
| 16:33.12 | Aiatek | :) |
| 16:33.24 | Aiatek | hehehehhehehe |
| 16:35.18 | pta200 | which dundi cli command allows you to verify that your server is adverising? I got the peers talking to each other but lookups don't return any results |
| 16:35.23 | *** join/#asterisk Pwn-BoFH (n=Coto@pc-86-7-239-201.cm.vtr.net) |
| 16:44.42 | Blackvel | how about my 2nd dial/busy macro try: http://pastebin.com/m1a47c583 |
| 16:47.04 | [TK]D-Fender | Blackvel: exten => 100,1,Macro(dial_with_busyhandling, ${PHONE_PEER1}) <-- stop adding whitespace, you're going to break things |
| 16:48.21 | [TK]D-Fender | Blackvel: Looks better, but I wouldn't bother with those constants you are referncing in your macro calls |
| 16:56.10 | *** join/#asterisk spiri (n=spiri@S01060002553240a8.vc.shawcable.net) |
| 16:56.15 | spiri | morning * |
| 16:57.15 | pta200 | Anybody know how to determine from the CLI what extensions dundi is advertising? |
| 16:57.27 | talirk81 | Is there any way to force an AGI to wait for a EXEC DIAL it has run to complete before continueing in the background ? basically i want to have a list of numbers to dial and have it Dial, see how long the dial tool (dialedtime - answeredtime) and then exit or go to the next if the caller picked up, but all through a single agi script. Not to have a Dial.php agi script for each dial attempt. |
| 16:58.23 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 16:59.32 | *** join/#asterisk WHYS (n=drumm@137.28.94.209) |
| 17:01.20 | *** join/#asterisk Failrar (n=Failrar@2001:888:1440:0:2a0:d1ff:fe4e:e802) |
| 17:02.35 | Blackvel | [TK]D-Fender: thanks. had it without constants and then I found a posting/suggestion on voip-info. thought your next statement would be "use constants" so I included before :) |
| 17:03.46 | [TK]D-Fender | Blackvel: Its only used one place so there is no point. |
| 17:04.05 | Blackvel | full agree |
| 17:04.05 | [TK]D-Fender | Blackvel: You kept asking how to make it smaller and more efficient and then went and added filler |
| 17:04.40 | Blackvel | one space really breaks? wow |
| 17:04.54 | *** join/#asterisk ingenius (n=alektro@186.136.6.218) |
| 17:05.18 | *** join/#asterisk xpot-mobile (n=james@144.35.21.73) |
| 17:05.20 | Blackvel | now I am happy that you point me to a good solution |
| 17:05.22 | Blackvel | really |
| 17:07.36 | Blackvel | <PROTECTED> |
| 17:07.50 | Blackvel | got that when I called in, let my phones ring and hangup (as caller) |
| 17:08.05 | Blackvel | does that mean it didn't abort normally/for good? |
| 17:08.24 | talirk81 | No ideas on a way to cause a AGI to wait for a dial to complete rather than continue on in the background while the dial is occuring? |
| 17:10.11 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 17:11.06 | [TK]D-Fender | Blackvel: I don't see your call attempt |
| 17:12.28 | *** part/#asterisk spiri (n=spiri@S01060002553240a8.vc.shawcable.net) |
| 17:14.02 | Blackvel | didn't paste.... |
| 17:14.43 | Blackvel | "exited non-zero" doesn't mean error? what is a good way to end a macro? s,n,Hangup is o.k in my case? |
| 17:15.34 | *** join/#asterisk RobertF (n=frederic@angela.tuxonet.be) |
| 17:15.36 | RobertF | Hello |
| 17:15.36 | [TK]D-Fender | Blackvel: means it aborted on the first line |
| 17:16.04 | Blackvel | first line of macro (where my dial cmd lives)? |
| 17:16.32 | RobertF | i'm setting up two sip account in sip.conf, if the number begins by 047 or 048 or 049, it uses the second account |
| 17:17.35 | [TK]D-Fender | RobertF: "it"? What is "it"? And what "number"? |
| 17:17.41 | [TK]D-Fender | blackyes, just like ti says. |
| 17:18.15 | s14ck | Aiatek, sup man |
| 17:18.28 | Aiatek | <s14ck> fine and you |
| 17:18.43 | RobertF | [TK]D-Fender: the number is a belgian mobile phone number. 0496XXXXXX |
| 17:19.06 | Blackvel | when the dial exists (caller hangs up) shouldn't it check for dialstatus CANCEL and move to lbl_hangup and Hangup()? |
| 17:19.08 | RobertF | [TK]D-Fender: i want to use the second account to call to Belgian mobile phone number |
| 17:19.31 | RobertF | The belgian mobile phone numbers begins: 047 - 048 - 049 |
| 17:20.10 | Blackvel | well probably Hangup does not help anymore when the line was hangup already ;) |
| 17:20.26 | Blackvel | i wonder if I am missing anything behind Dial |
| 17:20.37 | Blackvel | s/behind/after |
| 17:20.50 | tzafrir_laptop | http://lists.elastix.org/pipermail/beta-testers/2009-June/000555.html :-) |
| 17:24.21 | RobertF | [TK]D-Fender: can you help me? |
| 17:30.29 | [TK]D-Fender | RobertF: So far you aren't showing us a problem (ie: failed call). Pastebin is your friend... |
| 17:30.31 | [TK]D-Fender | ~pb |
| 17:30.31 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 17:33.02 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:34.14 | RobertF | ok |
| 17:37.54 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 17:38.27 | *** join/#asterisk layne (n=layne@ool-44c0048f.dyn.optonline.net) |
| 17:40.31 | RobertF | i search the problem.. |
| 17:40.46 | RobertF | i must to understand X,N,... |
| 17:44.17 | *** join/#asterisk tobias (n=tobias@216.27.28.176) |
| 17:49.15 | *** join/#asterisk dajhorn (n=chatzill@206.16.96.160) |
| 17:50.31 | *** join/#asterisk goodjoke (i=42c34605@gateway/web/ajax/mibbit.com/x-97f90209e6ccd372) |
| 17:53.35 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:53.35 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:54.41 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 17:57.08 | pif | hi, I have a 'playback' and then a 'queue' with music, how can I reduce the gap (silence) between the two? |
| 17:59.56 | *** join/#asterisk imcdona (n=imcdona@96.9.161.252) |
| 18:00.02 | *** join/#asterisk ruben23 (n=RPL@124.107.3.178) |
| 18:00.08 | imcdona | hello |
| 18:00.35 | talirk81 | is there a max limit to the length of a variables values in asterisk? |
| 18:00.36 | [TK]D-Fender | pif: Does the silense happen before Queue is called? |
| 18:00.57 | *** join/#asterisk tfrew (n=tfrew@office.neteasyinc.com) |
| 18:00.58 | [TK]D-Fender | talirk81: Yes, jsut like there is for dialplan lines. |
| 18:01.29 | Blackvel | what is a good way / best practice? |
| 18:01.42 | imcdona | I am pulling my hair out trying to get ondemand recording working. I have added both w and W to the dial command, and for some reason whenever I press *1 recording does not start. I have debugged DTMF to ensure asterisk is indeed getting the *1, have tried from callee and called party to no avail. Anyone have an idea's on what may be going wrong? |
| 18:02.04 | talirk81 | i cant find it oriely book , any idea where thats documented? |
| 18:02.19 | Blackvel | answer the line, ring the phone and if busy play prompt Playback(busy) and hangup? |
| 18:02.47 | imcdona | what module is the Ww option using? I am going to recompile it at this point |
| 18:03.41 | Blackvel | can Playback(busy) be used without charging units? when there is no answer my concept is to playback prompts and point the caller to an IVR menu (callback, voicemail, mobile interconnection, etc) |
| 18:03.45 | [TK]D-Fender | talirk81: Only in the source itself |
| 18:03.53 | [TK]D-Fender | talirk81: And its less than obvious |
| 18:04.10 | Blackvel | does it really suck to answer the call before and then follow that by Playback(busy) and Hangup? |
| 18:04.25 | *** join/#asterisk jlnt (n=lank1@70.255.193.190) |
| 18:04.51 | talirk81 | [TK]D-Fender, you dont happen to know a rought number on 1.4 |
| 18:05.33 | jlnt | hey guys, when our office in australia tries to make an outbound call using eyebeam to the office in the United States, it rings once then hangs up |
| 18:05.39 | [TK]D-Fender | talirk81: No, Someone mentioned it as they wrote a patch to circumvent this, but I didn't memorize the details |
| 18:05.40 | jlnt | any idea on the port forwarding on Asterisk Now |
| 18:06.18 | [TK]D-Fender | jlnt: By default it doesn't do port forwarding. Anything of that sort you expect the box itself to do remains a distro question. |
| 18:06.23 | jlnt | we have ports 5060 for SIP and 10000-20000 |
| 18:06.26 | jlnt | on the router |
| 18:06.42 | jlnt | it's Asterisk now |
| 18:07.21 | Blackvel | thats your rtp range? |
| 18:07.30 | jlnt | yeah |
| 18:08.20 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:08.20 | [TK]D-Fender | jlnt: read : |
| 18:08.22 | [TK]D-Fender | ~sipnat |
| 18:08.22 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:08.24 | [TK]D-Fender | ^^^^^^ |
| 18:09.05 | jlnt | I think I looked at this before |
| 18:09.17 | jlnt | but how do you change that configuration in sip.conf |
| 18:09.32 | jlnt | since, freepbx will just overwrite it with any changes made |
| 18:09.55 | KyleK | ~cepstral |
| 18:09.59 | KyleK | awwwwww |
| 18:10.31 | rue_mohr | so, if the echo can we have is the same for cards from 8 to 24 channels, and we have 6, I should be able to dial the tap count up quite a bit, yes? |
| 18:10.51 | Aiatek | <jlnt> it has a file sip_nat.conf |
| 18:11.04 | Aiatek | find it and edit it |
| 18:11.07 | jlnt | yeah |
| 18:11.09 | jlnt | let me find it |
| 18:11.16 | jlnt | I know it's in /etc/asterisk |
| 18:11.51 | jlnt | nothing there |
| 18:12.12 | [TK]D-Fender | jlnt: GUI's are not supported here, go ask in their channel |
| 18:12.24 | [TK]D-Fender | ~freepbx |
| 18:12.24 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:12.53 | jlnt | well, I am editing the sip_nat.conf file manually |
| 18:13.12 | KyleK | does digiums app_cepstral only work with the allison voice? |
| 18:15.24 | Qwell | res_cepstral? |
| 18:15.28 | dajhorn | Is there a conf option where I can make an agent login automatically by default? (I'm running 1.4.21.2~dfsg-3ubuntu2) |
| 18:15.59 | *** join/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
| 18:17.22 | [TK]D-Fender | dajhorn: No. |
| 18:17.56 | dajhorn | [TK]D-Fender: Could I script it? Can I force a login at the `asterisk -r` console? |
| 18:18.32 | [TK]D-Fender | dajhorn: type "help" at CLI. Do you see a command to do what you are looking for? |
| 18:20.08 | KyleK | hrm i thought it was app_, wewps |
| 18:20.25 | KyleK | oh yea it is res_cepstral |
| 18:20.28 | dajhorn | [TK]D-Fender: It has an 'agent logoff' command, but I don't see a corresponding 'agent logon' command. |
| 18:22.31 | *** join/#asterisk juanIMP (n=juan@200.71.41.254) |
| 18:22.44 | [TK]D-Fender | dajhorn: Then there is that answer. You need to execute dialplan to do the actions you're looking for it seems. Go think about how you would automate that for a while. |
| 18:24.22 | eppigy | DONDE ESTA |
| 18:25.05 | Blackvel | can I use Playback(busy) when snom phone is busy with no answer and option "don't answer the call" to avoid charged units? |
| 18:26.24 | pif | is there a way to playback while continuing the dialplan ? |
| 18:26.26 | *** join/#asterisk hi365 (n=hi365@94.159.177.112) |
| 18:32.37 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 18:32.49 | [TK]D-Fender | Blackvel: Playback = answer |
| 18:32.56 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:33.10 | [TK]D-Fender | pif: Background() |
| 18:34.21 | Blackvel | so even a Playback(busy) charges telco units or is the telco line not "really" answered? |
| 18:34.59 | Blackvel | for both my ivr sub-systems i do not care. but for a direct call it makes no sense to charge units |
| 18:35.09 | [TK]D-Fender | Blackvel: Playback = answer <-- what part of this is not clear? |
| 18:36.59 | pta200 | which dundi cli command allows you to verify that your server is adverising local extensions? I've got the peers talking to each other but lookups don't return any results |
| 18:39.03 | Blackvel | that is of course clear. the page http://www.voip-info.org/wiki/view/Asterisk+cmd+Busy seems to talk about some differnet solutions. don't get that |
| 18:39.12 | *** join/#asterisk superbeef (n=superbee@24.75.129.14) |
| 18:40.36 | superbeef | Is there any advantages to using a RT Kernel with Asterisk 1.2.x--latest |
| 18:41.37 | Blackvel | does Playtones(busy) and indications.conf helps better than Playback(busy) which apparently answers the line |
| 18:41.53 | *** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-e43a7677258884a4) |
| 18:43.09 | *** join/#asterisk tobias (n=tobias@216.27.28.176) |
| 18:47.42 | [TK]D-Fender | blatone = sound = audio = answer. |
| 18:47.58 | [TK]D-Fender | Blackvel: tone = sound = audio = answer. |
| 18:48.16 | [TK]D-Fender | superbeef: Nope. |
| 18:48.45 | superbeef | bummer! |
| 18:48.46 | [TK]D-Fender | Blackvel: "Busy" can work if you haven't answered the line and you are not dealing with analog PSTN |
| 18:49.00 | superbeef | <PROTECTED> |
| 18:49.16 | superbeef | [TK]D-Fender: thanks |
| 18:51.27 | [TK]D-Fender | superbeef: RT doesn't really matter to *. |
| 18:52.21 | *** part/#asterisk pta200 (n=paolo@69.38.173.162) |
| 18:52.38 | *** join/#asterisk buttons840 (n=buttons8@c-24-10-149-58.hsd1.ut.comcast.net) |
| 18:53.34 | buttons840 | Do you interface with the AMI the same way you would the AGI? |
| 18:54.18 | [TK]D-Fender | buttons840: No, they have nothing to do with wach other |
| 18:54.18 | [TK]D-Fender | buttons840: No, they have nothing to do with each other |
| 18:54.30 | eppigy | hi |
| 18:54.41 | eppigy | why are you angering [TK]D-Fender |
| 18:54.56 | buttons840 | the book never explains or gives example of how exactly the use the ami? |
| 18:55.13 | *** join/#asterisk mapoupier (n=marcandr@MTRLPQ19-1279392437.sdsl.bell.ca) |
| 18:55.26 | [TK]D-Fender | buttons840: Its a tcP connection, and the format of the command is documented, as is the response it gives. |
| 18:55.41 | eppigy | TCPIZZLE |
| 18:55.43 | [TK]D-Fender | buttons840: And there are plenty other docs whcih go into more deatil. |
| 18:56.45 | eppigy | There is a need for a ATFOT 3rd edition |
| 18:56.49 | eppigy | thought |
| 18:56.53 | eppigy | though |
| 18:58.23 | *** part/#asterisk ramindia (n=balajibh@202.63.96.10) |
| 18:58.36 | buttons840 | [TK]D-Fender, so these commands it gives as examples are tcp packets? |
| 18:58.50 | [TK]D-Fender | buttons840: Yes |
| 18:59.53 | buttons840 | dare i ask how to send and read tcp packets? I expect i'll have to look elsewhere ;) know any good sources off hand? |
| 19:00.10 | [TK]D-Fender | buttons840: Chapter 10 seems to talka alot about this. GO READ |
| 19:00.19 | buttons840 | i did |
| 19:00.46 | [TK]D-Fender | buttons840: And for that last part... this is a programming question. This is not #teachmeprogrammingidontknowanythingyet |
| 19:01.40 | buttons840 | indeed, as i said, i expect i'll have to look elsewhere, thanks |
| 19:02.03 | [TK]D-Fender | buttons840: Yup. I hope AMI is actually what you're looking for... |
| 19:02.59 | buttons840 | well, probably not, i doubt i will need it for my purposes, i was just wondering how one interfaces with it, and you answered that |
| 19:03.28 | [TK]D-Fender | buttons840: That I did. |
| 19:03.31 | buttons840 | i'll know where to look in the future if i need it |
| 19:03.42 | *** join/#asterisk p4tr0p1 (n=jam@201-34-141-194.fnsce704.e.brasiltelecom.net.br) |
| 19:03.55 | p4tr0p1 | hi everyone |
| 19:04.44 | telnettech | have a networking question.......if i were to change the DHCP server, would my phones renew their leases immediately with the new DHCP server or will I have to wait until the lease expires? |
| 19:04.54 | p4tr0p1 | I have a asterisk server, using Zap with a Digium card |
| 19:05.12 | telnettech | for the phones? |
| 19:05.28 | *** join/#asterisk imcdona (n=imcdona@96.9.161.252) |
| 19:05.56 | [TK]D-Fender | telnettech: I've never heard of a "DHCP server online, get new IP's now" function before. |
| 19:05.56 | Nugget | telnet is eeeeeeevil! |
| 19:06.06 | p4tr0p1 | everything works perfectly, but my costumer wants to "hide" the line ID when calling pstn |
| 19:06.27 | [TK]D-Fender | p4tr0p1: Depends what card, and what signalling |
| 19:06.36 | p4tr0p1 | Signaling R2 |
| 19:07.16 | p4tr0p1 | card Digium Wildcard TE110P T1/E1 |
| 19:08.05 | telnettech | TK: Thats what im asking. when the phones get the broadcast message from DHCP server and "sees" that the MAC address is different for the DHCP server than what they have in memory, If they will automatically renew or if I have to go and restart every phone to get a new IP address from new DHCP Server? |
| 19:09.10 | [TK]D-Fender | p4tr0p1: I don't know what R2 supports exactly, but go through : "core show application setcallerpres" |
| 19:09.38 | [TK]D-Fender | telnettech: I've never heard a DHCP server "broadcast" before. |
| 19:09.53 | telnettech | TK: ok thanks |
| 19:10.54 | moy | p4tr0p1: if by line id you mean callerid, hidecallerid should work in openr2 (don't remember about unicall, but I think it should work as well), however most of the time the telco does not require the caller id anyways (if you're connected to the telco) |
| 19:11.46 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-182-101.lns10.mel4.internode.on.net) |
| 19:11.53 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 19:11.58 | *** join/#asterisk Failrar (n=Failrar@2001:888:1440:0:2a0:d1ff:fe4e:e802) |
| 19:12.28 | moy | but if you're using chan_dgv you should ask digivoice then |
| 19:13.04 | superbeef | telnettech: They'll wait to renew their leases at their half time... It'll be best to configure yoru existing dhcp server to shorter leases a few days in advance |
| 19:15.03 | telnettech | superbeef: thanks. we are going to restart the data switches 1 floor at a time to get new ip addresses....we were looking for a way to keep the bright light from happening when the phones reboot. This i going to be done at 3 or 4 am when hotel guests are sleeping. |
| 19:17.16 | p4tr0p1 | moy, Yes, I'm connected to telco, I've talked to them and they saw that it doesn't metter if the callerid is sent or not |
| 19:17.48 | moy | yep, because they know who you are :) |
| 19:18.23 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 19:18.55 | p4tr0p1 | [TK]D-Fender, moy: tks folks, I'll test some options and back here =) |
| 19:20.32 | Blackvel | [TK]D-Fender: thank you very much for pointing this out for me! |
| 19:20.52 | Blackvel | have a good evening and happy programming |
| 19:22.24 | jlnt | what's the command to upgrade the asterisknow? |
| 19:22.37 | jlnt | wget what? |
| 19:23.40 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 19:23.42 | *** join/#asterisk layne (n=layne@ool-44c0048f.dyn.optonline.net) |
| 19:25.29 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 19:25.55 | e0n` | So, any recommendations with an auto dialer that can integrate with asterisk and is 'affordable' not to put to fine of a word on it |
| 19:26.50 | e0n` | Can I do them with the outgoing spool if I write my own? |
| 19:27.18 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 19:27.21 | bmoraca | so does anyone have any experience or feedback with the SPA8000-G1? seems like a decent device, but i'd like to know if I'm going to have any surprises before I start using them...any feedback good or bad? |
| 19:28.34 | *** join/#asterisk Micc (n=dotirc@c-76-121-255-52.hsd1.wa.comcast.net) |
| 19:28.36 | kaldemar | e0n`: manager interface originate or spool if you write your own. |
| 19:32.21 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:34.32 | [TK]D-Fender | jlnt: Upgrade what? |
| 19:35.19 | [TK]D-Fender | bmoraca: Pretty decent. Largly like 4 ATA's bolted together |
| 19:36.06 | KyleK | Qwell: are you familiar with res_cepstral? I'm not clear on why it requires a port license but apparently running the swift engine directly just requires a voice license. |
| 19:37.16 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-3177291277d7be07) |
| 19:37.21 | *** join/#asterisk NirS (n=NirS@80.250.159.240) |
| 19:37.28 | NirS | good evening everybody |
| 19:37.37 | NirS | or good morning, wherever the case may be |
| 19:40.00 | *** join/#asterisk Blackvel (n=blackvel@dslb-088-065-105-077.pools.arcor-ip.net) |
| 19:40.06 | Blackvel | i am back. it's not working :) |
| 19:40.33 | Blackvel | shouldn't there be a busy.alaw/busy.ulaw default sound file? |
| 19:40.48 | *** join/#asterisk voipnet-tech (n=voipnett@rrcs-24-39-187-146.nys.biz.rr.com) |
| 19:41.28 | [TK]D-Fender | Blackvel: Says who? |
| 19:41.47 | Blackvel | on wiki and stuff they all reference/suggest Playback(busy) |
| 19:42.19 | [TK]D-Fender | Blackvel: jsut because the WIKI shows you a sample indicationg a file, who says its an * provided one? |
| 19:42.41 | [TK]D-Fender | Blackvel: Keep in mind the WIKI is poorly maintained and often by schmucks |
| 19:42.49 | Blackvel | well just thought so :) |
| 19:42.51 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 19:45.34 | voipnet-tech | Hello all! I'm working on a new app today and looking for some existing scripts/dialplan to learn from. I need it to do the following. 1) User enters a phone number in a browser, 2) Web server initiates action thru Asterisk Manager 3) Asterisk first must dial a phone number of an IVR, pause, then enter a 6 digit PIN#. 4) When the PIN is entered, Asterisk originates the call to the phone number entered from the browser and jo |
| 19:45.46 | Blackvel | i am a bit confused anyways.... |
| 19:46.11 | *** join/#asterisk rgsteele||work (n=rgsteele@75-147-74-137-Philadelphia.hfc.comcastbusiness.net) |
| 19:46.24 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 19:46.42 | Blackvel | leifmadsen suggested using outer Dial with Local channel construct and answer the local channels to play busy signal |
| 19:46.43 | voipnet-tech | PS: I seem to be having difficulty finding how to originate the 2 calls and get them joined together |
| 19:46.57 | rgsteele||work | I'm in the process of migrating from a 1.2 box to a 1.6 box, and I was wondering what the purpose of res_pgsql.conf was - it's existant in my 1.6 config dir and asterisk appears to really want to connect to a database whose credentials are stored there. |
| 19:47.25 | Blackvel | with my 1st dial (Local/peer1&Local/peer2) I check for BUSY dailstatus and play a custom busy prompt file |
| 19:47.29 | *** join/#asterisk kram (n=markster@nat/digium/x-cc3228bde8b4c6f9) |
| 19:47.36 | Blackvel | and i don't get there |
| 19:48.36 | *** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-edaa36c9230d2e51) |
| 19:48.38 | Blackvel | <PROTECTED> |
| 19:49.14 | Blackvel | but that means that there is no way that the inner dial (local channel) is able to check for BUSY state and return with BUSY? |
| 19:49.45 | Blackvel | i can not see a benefit having to handle the busy per local channel (and playback my prompt) |
| 19:50.03 | rgsteele||work | Looks like it's trying to use a realtime connection, which is weird because I've not explicitly configured that. |
| 19:50.16 | Blackvel | wouldn't it be better to have the option to return to outer-dial BUSY dialstatus so the outer BUSY-check win's? |
| 19:50.38 | [TK]D-Fender | voipnet-tech: AMI Originate or Call files. take your pick. |
| 19:51.41 | Blackvel | i don't have any idea when the outer-dial BUSY even win's (as my local channels will get answered all the time as they have to do BUSY checking to) |
| 19:52.36 | nkohh | doritos do not go with coffee |
| 19:52.36 | [TK]D-Fender | Blackvel: And you aren't showing us anything |
| 19:52.38 | nkohh | fyi |
| 19:52.53 | [TK]D-Fender | nkohh: Sure they do. coffe goes with everything |
| 19:53.04 | nkohh | [TK]D-Fender: maybe it's just the hot wing/blue cheese doritos |
| 19:54.45 | Blackvel | oh my dear now I got it: i can't even read the wiki.... confused "Playtones(busy)" with Playback(busy) ;) |
| 19:55.15 | [TK]D-Fender | Blackvel: Both will answer BTW |
| 19:55.27 | voipnet-tech | [TK]D-Fender, or anyone else, I've ready about the call files and AMI originate, and understand what they're doing, but they are requiring an extension number to call first before dialing the numbers... This * has no phones registered to it so i wasn't sure how that would work |
| 19:55.59 | Blackvel | no wonder I didn't find busy.alaw :) |
| 19:56.10 | [TK]D-Fender | voipnet-tech: It dials the Channel: you provide it and then dumps you into the dialplan where you tell it to. Y don't need ANYTHING connected to * at all to do this. |
| 19:56.21 | [TK]D-Fender | You* |
| 19:56.47 | voipnet-tech | ok i think i understand, here goes nothing |
| 19:57.01 | bmoraca | best way to learn is to fuck up a few times |
| 19:57.50 | *** join/#asterisk layne (n=layne@ool-44c0048f.dyn.optonline.net) |
| 19:59.37 | [TK]D-Fender | bmoraca: No, best way would be to succeed after a few screwups so you can understand what makes it work where the other attempts failed :) |
| 20:00.04 | [TK]D-Fender | bmoraca: Just fucking up doesn't teach you how, it only teeaches you how not to :) |
| 20:00.20 | bmoraca | yeah...that second part might be useful...but there's a lot to be gained from the knowledge of how NOT to do somnething |
| 20:02.51 | [TK]D-Fender | bmyup, Edison mastered that all right... |
| 20:03.42 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 20:03.45 | *** join/#asterisk viq_ (n=viq@unaffiliated/viq) |
| 20:05.06 | RobertF | [TK]D-Fender: I solve the problem.. |
| 20:11.04 | jplank | wow - boss just called me with $3k worth of unknown charges to canada, thought for sure I was hacked |
| 20:11.25 | jplank | that god it was just a miss programming on my part that billed it to the wrong telephone number :/ |
| 20:11.53 | *** join/#asterisk Pwn-BoFH (n=Coto@pc-86-7-239-201.cm.vtr.net) |
| 20:14.13 | *** join/#asterisk viq_ (n=viq@unaffiliated/viq) |
| 20:16.06 | *** join/#asterisk voxter (n=voxter@200.122.188.222) |
| 20:16.51 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 20:18.09 | *** join/#asterisk buttons840 (n=buttons8@c-24-10-149-58.hsd1.ut.comcast.net) |
| 20:23.47 | *** join/#asterisk ingenius (n=alektro@host176.190-230-72.telecom.net.ar) |
| 20:24.03 | [TK]D-Fender | checkout time, bbiab |
| 20:25.34 | ruben23 | hi |
| 20:26.01 | ruben23 | [TK]D-Fender: the voip im setting last day was not working |
| 20:26.07 | ruben23 | i cant get it through |
| 20:26.12 | ruben23 | :-( |
| 20:33.09 | *** join/#asterisk keith4 (n=keith@unaffiliated/keith4) |
| 20:34.01 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 20:36.33 | *** join/#asterisk keith4 (n=keith@unaffiliated/keith4) |
| 20:42.53 | *** part/#asterisk superbeef (n=superbee@24.75.129.14) |
| 20:47.27 | pfn | hmm, looks like my silent voicemail patch works, yay... either that or everything is getting discarded ;-) |
| 20:48.46 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 20:54.26 | KyleK | so leave a message? |
| 20:54.39 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 20:59.31 | timeshell_atwork | [TK]D-Fender xipi should have simply set up an Incoming Rule |
| 20:59.39 | timeshell_atwork | in the gui that is |
| 21:00.05 | Blackvel | how to pass the control back from answered local channel to originating (outer) dial? is this even possible when the dial of a local channel has been answered? |
| 21:00.10 | timeshell_atwork | It looks to me like he had never set one up. |
| 21:00.30 | timeshell_atwork | And... [TK]D-Fender is not here |
| 21:00.32 | timeshell_atwork | :( |
| 21:01.14 | timeshell_atwork | Blackvel Say what? |
| 21:02.02 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:02.29 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:05.50 | pfn | KyleK, too lazy to try right now |
| 21:06.03 | KyleK | hahaha |
| 21:06.06 | KyleK | whats your # ;) |
| 21:06.34 | pfn | it's ok, I'll just wait |
| 21:07.37 | *** join/#asterisk BadHAL (n=nn@70.99.106.38) |
| 21:08.19 | Katty | eppigy: :< |
| 21:08.25 | Katty | eppigy: hungries. |
| 21:10.00 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 21:11.47 | ariel_ | Katty: I am, so I just started on a Granola bar, Nature Valley's Oats'N Honey....good really |
| 21:12.05 | eppigy | Katty: me2 |
| 21:12.18 | eppigy | ariel_: that is just cruel |
| 21:12.26 | ariel_ | why |
| 21:12.34 | eppigy | here we are starving |
| 21:12.53 | eppigy | and you are all "oh i have this delicious nature valley granola bar" |
| 21:12.53 | ariel_ | it's only 50 cents in the vending machine here |
| 21:13.15 | *** join/#asterisk angryuser_ (n=angryuse@90-156-167-83.reverse.alphalink.fr) |
| 21:20.12 | bmoraca | VMWare ESX is wonderful |
| 21:20.57 | timeshell_atwork | awk_r ping |
| 21:22.45 | awk_r | redirects timeshell_atwork to #asterisk-gui |
| 21:25.14 | *** join/#asterisk Aiatek (n=munoz@adsl-54-126.tricom.net) |
| 21:26.20 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 21:27.11 | *** join/#asterisk RoPBX (n=nickserv@200.93.34.175) |
| 21:27.18 | RoPBX | hi all |
| 21:28.13 | *** join/#asterisk blkry (n=blkry@64.147.222.130) |
| 21:31.44 | RoPBX | [TK]D-Fender, please I ask you few days ago about how to process parked calls, and you told me to use asterisk -rx show channel from an AGI |
| 21:32.19 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 21:32.38 | RoPBX | i already did it, but when I call the AGI the channel is dead, and if i call the AGI before the ParkedCall, the channel is not bridget and i cant get the info |
| 21:33.07 | RoPBX | can you please help me? |
| 21:34.17 | *** join/#asterisk [jmc] (n=John@93-45-217-68.ip104.fastwebnet.it) |
| 21:34.36 | [jmc] | hi guys ;) |
| 21:36.02 | [jmc] | I don't know if this is the right place to ask that |
| 21:36.33 | [jmc] | but I'd like to hear some advices from you |
| 21:36.46 | [jmc] | because I'll have to propose some choices for an Asterisk setup |
| 21:37.02 | Joel | huh? |
| 21:37.22 | [jmc] | and I tried for a little while AsteriskNOW and Trixbox |
| 21:37.35 | [TK]D-Fender | [jmc]: Find your point, and then take a flying leap at it :p |
| 21:37.44 | [jmc] | lol |
| 21:37.53 | [jmc] | oh, right down to the core :P |
| 21:38.23 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 21:38.31 | [jmc] | what Asterisk distribution would you recommend as a good choice? |
| 21:38.54 | [TK]D-Fender | [jmc]: Any that you can comfortably manage and install * and its dependencies on |
| 21:39.14 | [jmc] | thanks :P |
| 21:39.19 | grndslm | even ubuntu?!? |
| 21:39.23 | [jmc] | is there one that offers support (commercial)? |
| 21:39.48 | [jmc] | or is it a Digium task? |
| 21:40.12 | [TK]D-Fender | [jmc]: Support of what? |
| 21:40.33 | [jmc] | technical support |
| 21:40.36 | [TK]D-Fender | grndslm: Yes, even Ubuntu. |
| 21:40.41 | [TK]D-Fender | [jmc]: OF WHAT? |
| 21:40.46 | grndslm | support of answering the phone? |
| 21:40.53 | [jmc] | in case they have problems of any kind, I don't know :D |
| 21:41.13 | [TK]D-Fender | [jmc]: WHAT is being supported? By WHOM? |
| 21:41.37 | [jmc] | Asterisk |
| 21:42.06 | [TK]D-Fender | [jmc]: You can get paid support from Digium for that |
| 21:42.49 | [jmc] | ok, good |
| 21:43.00 | grndslm | or free support from TK!! |
| 21:43.04 | grndslm | :P |
| 21:43.10 | [jmc] | right :P |
| 21:43.36 | [jmc] | or maybe he's working for Digium |
| 21:43.45 | [jmc] | he'll be asking for money after some help :D |
| 21:44.18 | grndslm | if TK works for Digium, then I'm Sarah Palin |
| 21:45.48 | [jmc] | and I'm John McCain |
| 21:45.52 | [TK]D-Fender | [jmc]: nope.... |
| 21:46.00 | [TK]D-Fender | [jmc]: I do PRE-PAID support :) |
| 21:46.07 | [jmc] | (indeed, that's what "jmc" stands for :-|) |
| 21:46.14 | [jmc] | :D |
| 21:46.58 | [TK]D-Fender | [jmc]: With all the problems you come in with I figured it was "jacked my computer" :p |
| 21:47.17 | [jmc] | çP |
| 21:47.18 | [jmc] | :P |
| 21:47.31 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 21:48.24 | [jmc] | I'm an Asterisk novice, but I think I use the computer just fine |
| 21:51.08 | [jmc] | yet I know I made a fool of myself with all my problems in this channel ;) |
| 21:53.19 | Katty | eppigy: hunnnngryyyyyyyyyyyyyy /cry |
| 21:54.07 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 21:54.26 | freenose | Will the asterisk book be good enough for 1.6.1? |
| 21:55.04 | [jmc] | bye. |
| 21:58.13 | nkohh | makes a fool of himself just about everywhere |
| 21:58.15 | [TK]D-Fender | freenose: Things have changed, there are docs in the tarball. Read up |
| 22:09.06 | *** join/#asterisk squaretro (n=scott@elyria-max3-cs-16.dialne.bright.net) |
| 22:09.24 | squaretro | is it possible to add cic to sip invite? |
| 22:10.15 | squaretro | all my LD is being blocked because there is no cic |
| 22:10.44 | squaretro | i guess ive been bypassing toll for like 5 months and just found out today |
| 22:11.40 | *** part/#asterisk tfrew (n=tfrew@office.neteasyinc.com) |
| 22:14.15 | freenose | [TK]D-Fender: ok |
| 22:15.08 | squaretro | from what i understand i need to add cic to my invite |
| 22:15.19 | squaretro | if its possible |
| 22:17.21 | [TK]D-Fender | squaretro: Perhaps you could explain what CIC is supposed to look like... |
| 22:18.11 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.71) |
| 22:18.13 | ruben23 | hi |
| 22:18.13 | squaretro | oh m sorry |
| 22:19.51 | squaretro | NXXNXXXXXX;cic=0333@xxx.xxx.xxx.xxx |
| 22:20.30 | squaretro | supposed to be in invite |
| 22:20.56 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 22:23.22 | [TK]D-Fender | squaretro: Sounds like it should be part of what you dial then |
| 22:24.42 | squaretro | i c |
| 22:25.00 | squaretro | sorry im kinda new to this |
| 22:26.53 | *** join/#asterisk bijit (i=1000@190.241.15.48) |
| 22:26.59 | squaretro | any suggestions on a rule or dial pattern? |
| 22:27.31 | bijit | is there a way I could monitor on CLI a single extension? |
| 22:27.53 | [TK]D-Fender | squaretro: It isn't the pattern on the exten, its the EXTRA chars you put in the Dial command itself |
| 22:28.05 | [TK]D-Fender | bijit: No. |
| 22:28.35 | squaretro | i c |
| 22:29.15 | squaretro | is there any way you could give me an example to follow and try? |
| 22:29.22 | bijit | ty [TK]D-Fender |
| 22:29.55 | [TK]D-Fender | squaretro: Go look at your dial command and shove the ";cic=0333" right after the number you;re dialing |
| 22:30.41 | squaretro | plz forgive my ignorance |
| 22:30.51 | *** join/#asterisk imcdona (n=imcdona@96.9.161.252) |
| 22:31.10 | ruben23 | can anyone guide my sip trun config on my asterisk try serveral times no success. http://pastebin.com/m9a447ee |
| 22:31.14 | [TK]D-Fender | squaretro: DIAL <--- you know, where you CALL out your provider. Its your dialplan. Go in and LOOK where you place the call |
| 22:32.04 | squaretro | yes i kno i was talking about not knowing what im doing |
| 22:32.10 | [TK]D-Fender | ruben23: exten => _Z.,1,Dial(SIP/grnvoip/990131501${EXTEN}) <-- I told you this DAYS ago |
| 22:32.54 | ruben23 | <PROTECTED> |
| 22:33.19 | [TK]D-Fender | ruben23: And like usual you never show us the problem and what you're doing so we can't tell. |
| 22:34.13 | *** join/#asterisk `paul (n=kutimoy@121.97.99.151) |
| 22:34.52 | `paul | do you guys know any good toll free numbers provider in the US. i want to buy in bulk.... :) |
| 22:35.34 | ruben23 | [TK]D-Fender: this is the output of my test call http://pastebin.com/md27452e |
| 22:35.53 | bijit | ~isp |
| 22:35.53 | infobot | rumour has it, isp is (Internet Service Provider) An ISP provides Internet access to people or corporations. ISPs generally have pools of modems awaiting dial-up connections. Smaller ISPs buy bandwidth from larger ISPs. There are two basic ways of connecting: one is to get a PPP or Slip account; the other is to get a UNIX shell account. |
| 22:36.42 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 22:43.23 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 22:44.46 | *** join/#asterisk fskie_ (n=fskie@unaffiliated/fskie) |
| 22:44.47 | *** join/#asterisk comfrey (n=comfrey@66.166.226.6) |
| 22:45.21 | comfrey | i am having dificulties getting wanpipe going... |
| 22:45.34 | comfrey | all the different versions and compatabilities |
| 22:45.49 | comfrey | latest is /usr/src/wanpipe-3.4.1/kdrvtmp/sdla_tdmv.c:1105: error: âstruct dahdi_spanâ has no member named âechocanâ |
| 22:46.19 | comfrey | it compiles ok on dahdi 2.1 but not 2.2 |
| 22:46.47 | comfrey | can someone point me to a set of compatible versions for 2.6.26 kernel? |
| 22:47.14 | comfrey | i am experiancing frustration |
| 22:49.11 | comfrey | any ideas? |
| 22:51.55 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 22:55.40 | comfrey | ok, well... i will try again with the sangoma tools and the 1.4.25 and dahdi 2.1.0.x... |
| 22:56.23 | [TK]D-Fender | ruben23: AndrubWell you arn't even specifying a username in there. |
| 22:56.37 | [TK]D-Fender | ruben23: Also you aren't looking at the SIP DEBUG for the call. |
| 22:58.52 | *** part/#asterisk fskie_ (n=fskie@unaffiliated/fskie) |
| 23:00.13 | *** join/#asterisk imcdona (n=imcdona@c-24-19-203-112.hsd1.wa.comcast.net) |
| 23:00.43 | [TK]D-Fender | ruben23: AND you have no secret. You have NO account information in that peer entry at all to identify your aoount with them. |
| 23:01.40 | ruben23 | [TK]D-Fender: ok im getting all the stuff now and test it again |
| 23:01.48 | ruben23 | tahnks for the guidance |
| 23:07.00 | comfrey | ok, about to build wanpipe 3.4.1 |
| 23:07.36 | comfrey | building against dhadi 2.1.0.4 |
| 23:15.02 | comfrey | Configuring interfaces: w1g1 w1g1: ERROR while getting interface flags: No such device |
| 23:15.24 | comfrey | so i think the driver is not loading properly |
| 23:15.31 | comfrey | sangoma a200 |
| 23:22.54 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 23:33.33 | comfrey | migrating to dahdi is a big headache |
| 23:34.01 | [TK]D-Fender | comfrey: Your last error was clearly a purely wanpip issue |
| 23:35.19 | *** join/#asterisk rnovotny22 (n=rnovotny@71-220-122-121.mpls.qwest.net) |
| 23:36.03 | *** join/#asterisk Tar-Get (n=Tar-Get@83.101.36.144) |
| 23:37.18 | Tar-Get | hi, can someone help me? I get this error: [Jun 18 01:27:12] ERROR[11210] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data |
| 23:39.43 | rnovotny22 | Good evening. Need a little help. I am not able to receive sip calls from bv, but am able to make outgoing calls. |
| 23:43.17 | *** part/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 23:47.05 | comfrey | [ 1278.865586] ERROR: wanpipe1: No FXO/FXS modules are found! |
| 23:47.17 | comfrey | ok, so there are no modules found... |
| 23:47.46 | comfrey | <PROTECTED> |
| 23:48.05 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 23:48.08 | comfrey | i have a a200 with 6 fxo and 2 fxs modules... |
| 23:48.24 | comfrey | shouldn't they show up under hwprobe? |
| 23:48.39 | alrs | comfrey: you have the molex power connector connected? |
| 23:48.50 | comfrey | the 4 pin... yes |
| 23:49.37 | *** join/#asterisk juanIMP (n=juan@200.26.152.222) |
| 23:50.08 | alrs | comfrey: I haven't touched one of those cards in two years, but usually I saw that problem when there was something wrong with the power connector |
| 23:50.32 | alrs | comfrey: Sangoma is pretty helpful, if you have the means for them to ssh in to the system they'll probably be able to help you tomorrow morning |
| 23:52.11 | alrs | comfrey: you might try pulling the two halves of the card apart and then reseating everything |
| 23:53.31 | comfrey | alrs: messing with things now.. |
| 23:55.09 | comfrey | alrs: are there any jumpers that need to be set? |
| 23:55.48 | *** part/#asterisk korihor (n=korihor@190.72.254.245) |
| 23:58.36 | jlnt | hey after adding my voicepulse routes, it's stuck on Please wait, reloading.. |
| 23:59.16 | *** join/#asterisk Pwn-BoFH (n=Coto@pc-86-7-239-201.cm.vtr.net) |