| 00:00.01 | TSM | have you seen that howler now make realy high powered transcoder cards that blow digium out the water |
| 00:01.16 | rbd_ | citywok, we were able to get about 450-550 sessions on the boxes we were testing on (but they are desktop-class machines) |
| 00:01.36 | citywok | lol -- get a quad core and you dont have to worry about splitting boxes :) |
| 00:05.49 | Tech_Travis | Are there any phones available that can register with 2 separate * servers? I'd like a home phone that registers the first line with an * box at the office and have a second line register with my home * server. |
| 00:06.05 | citywok | Tech_Travis: pretty much any 2 line phone will do that |
| 00:07.14 | Tech_Travis | citywok: I've got a Cisco 7960 at the office and was thinking about getting one for home but wasn't seeing how to actually configure it that way. I guess I need to keep digging. Thanks. |
| 00:07.26 | citywok | i'm sorry for you |
| 00:08.01 | Tech_Travis | citywok: I was wondering if I needed to get an Aastra or Grandstream to make it work. |
| 00:08.19 | citywok | grandstreams sucketh mucheth |
| 00:08.24 | citywok | i love my aastras |
| 00:08.44 | Tech_Travis | citywok: any model in particular? |
| 00:09.01 | ChannelZ | Tech_Travis: doesn't your phone's web interface list separate tabs for line 1, 2, ec |
| 00:09.26 | citywok | i like my 6757i |
| 00:09.34 | citywok | (all 50 of them, except for the one that bricked) |
| 00:10.39 | Tech_Travis | ChannelZ: I didn't know the phone even had a web interface. I was trying to set it up through the SIP<macaddress>.cnf file. |
| 00:11.46 | *** join/#asterisk dlynes (n=dlynes@bas6-hamilton14-1167992947.dsl.bell.ca) |
| 00:12.05 | ChannelZ | I'm guessing it does, all these cisco/linksys/sipura phones are built on the same ground. I don't know anything about your specific phone but I'm assuming... |
| 00:12.07 | TSM | anyone here have a blackberry, do you think they will ever put a SIP stack in one for the enterprise customers |
| 00:13.08 | dlynes | Does anyone know anything about the 'm' or 'p(...)' options to Directory? |
| 00:14.18 | dlynes | Just trying to figure out why 'p(...)' is an advantage to cell phone callers, and how Directory() generates the menu that it's told to use with the 'm' option |
| 00:14.44 | bmoraca | ChannelZ, the Cisco 7900 phones and the Sipura-based Linksys/Cisco phones are VERY, VERY different. |
| 00:15.04 | bmoraca | ChannelZ, the 7940/7960 phones even more, as Cisco bought those models from someone else. |
| 00:15.22 | ChannelZ | I guess they just stole the plastic molds |
| 00:15.26 | Tech_Travis | ChannelZ: It may have it under a SCCP or MGCP but I'm running SIP 8.12 and it doesn't seem to have a web interface. |
| 00:16.00 | dlynes | ChannelZ: that's like saying Aastra phones and Nortel phones are the same, just because they have the same mold |
| 00:16.08 | bmoraca | Tech_Travis, what phone are you using? |
| 00:18.38 | Tech_Travis | bmoraca: I'm using a 7960G with SIP 8.12 image on * 1.6.0.19 SIP. |
| 00:18.44 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:e5ef:294a:6f1a:5546) |
| 00:18.59 | bmoraca | Tech_Travis, that should be able to register with multiple proxies no problem. pastebin your config files |
| 00:20.02 | Tech_Travis | bmoraca: sure, give me a minute to grab them. |
| 00:21.12 | bmoraca | wow...netflix's stock tanked today |
| 00:21.52 | ChannelZ | buy low sell high |
| 00:22.21 | bmoraca | unless you think they'll go lower...which they might |
| 00:22.42 | ChannelZ | It must have been all those DVDs I stole from them |
| 00:22.51 | bmoraca | lol |
| 00:26.37 | jblack | Russia is gonna adjust Apophis' orbit |
| 00:28.41 | Tech_Travis | bmoraca: http://pastebin.ca/1732491 <---- SIPDefault.cnf and SIP<macaddress>.cnf of the 7960 |
| 00:29.17 | bmoraca | ugly file, heh |
| 00:30.09 | Tech_Travis | bmoraca: The whole thing or just certain parts? |
| 00:30.19 | bmoraca | the whole thing. i hate comments :P |
| 00:31.18 | Tech_Travis | bmoraca: I figured I'd be nice so when I get a new job and the next poor bastard has to keep things working it won't be too bad for them :) |
| 00:33.34 | bmoraca | Tech_Travis, here's an exhaustive list of all 7940/7960 phone configuration options, valid settings, and what they do. i believe you'll want to set multiple proxy_addresses |
| 00:34.08 | MikeJ | I'll ask again sense there were no takers first time |
| 00:34.16 | MikeJ | anyone have a cisco/linksys 501 phone handy |
| 00:34.53 | bmoraca | Tech_Travis, yes, each proxyx_address corresponds to each linex_ setting. so, to set a different proxy for line 2, you'd add a setting for proxy2_address. if you do not specify a proxyx for each linex, it defaults to proxy1_address. |
| 00:35.00 | bmoraca | Tech_Travis, linky: http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7960g_7940g/sip/8_0/english/administration/guide/sipaxd80.html |
| 00:35.19 | bmoraca | MikeJ, i didn't know Cisco/Linksys made a 501 model phone |
| 00:35.38 | *** join/#asterisk cesar_CR (i=cesar@201.201.41.242) |
| 00:36.18 | Tech_Travis | bmoraca: OOOH. I saw I could set multiple proxies but I thought that was for failover in case the first was not reachable. |
| 00:36.32 | MikeJ | http://www.ciscosystems.or.at/en/US/prod/collateral/voicesw/ps6788/phones/ps10499/data_sheet_c78-548558.html |
| 00:36.57 | MikeJ | bmoraca: ^^ that one |
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| 00:37.29 | *** part/#asterisk cesar_CR (i=cesar@201.201.41.242) |
| 00:37.32 | MikeJ | its simialr to the 525g |
| 00:37.36 | MikeJ | but no screen |
| 00:37.45 | *** join/#asterisk cesar_CR (i=cesar@201.201.41.242) |
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| 00:38.23 | bmoraca | MikeJ, nope, i don't like the SPA phones. |
| 00:38.29 | Tech_Travis | bmoraca: The link looks very handy. Thanks a lot for your help. |
| 00:38.55 | bmoraca | Tech_Travis, no problem...finding information on the Cisco 7900 phones can be a bear |
| 00:39.01 | MikeJ | bmoraca: I don't like them either, I just need to find out something about them :D |
| 00:41.54 | mykhyggz | I'm wondering if I can tell dahdi to use 'echo' from kernel just by editing a text file. |
| 00:42.03 | *** part/#asterisk nny (n=scott@64.203.239.83) |
| 00:48.18 | bmoraca | woo! gohome time and another short week. |
| 00:48.35 | bmoraca | 4 more days of doin nothing |
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| 00:59.17 | *** join/#asterisk beters (n=beters@c-68-54-100-46.hsd1.ms.comcast.net) |
| 00:59.44 | beters | how can i get the caller id of the imcoming call in my dial plan? |
| 00:59.58 | beters | is there a variable for that? |
| 01:00.11 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 01:01.27 | dlynes | beters: CALLERID(num), CALLERID(name) |
| 01:01.38 | Micc | does 1.6 have transfer ring back feature? |
| 01:02.16 | Micc | If the transfer fails can I have the tansferer get a call back with the caller? |
| 01:02.21 | beters | hmmm, do you know what the value that it returns looks like? |
| 01:02.33 | beters | im trying to pass it as a vriable and put it in a database |
| 01:02.38 | beters | but it fails on it |
| 01:03.04 | beters | does it have any quotes in it or anything? |
| 01:09.12 | *** part/#asterisk Tech_Travis (n=Travis@mail.techglia.com) |
| 01:09.23 | beters | well, that really shouldnt matter |
| 01:09.36 | beters | that variable is coming up empty it seems |
| 01:09.45 | beters | which doesnt make sense |
| 01:09.51 | *** part/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 01:10.00 | beters | is there anything else i have to do other than just use that variable? |
| 01:11.37 | [TK]D-Fender | beters: its a FUNCTION go read your dialplan basics on thenm |
| 01:12.07 | beters | will do |
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| 01:32.40 | Micc | I read somewhere that in 1.6 theres an option to have blind xfers ring back if they fail. |
| 01:32.55 | Micc | But I can only find a patch to do it. |
| 01:33.39 | Micc | The patch was from 2006, so that can't be what they were talking about for 1.6. |
| 01:38.47 | Katty | hello |
| 01:40.02 | beters | can i not use an "&" for a link in a dialplan? |
| 01:40.27 | Katty | SIP/100&SIP/101, etc |
| 01:40.34 | Katty | you can use it like that. |
| 01:40.48 | beters | crap |
| 01:41.01 | beters | im calling a link with lynx that needs two variables |
| 01:41.03 | voipmonk | doesnt always work like it used to - |
| 01:41.17 | Katty | beters: so far i don't see a problem. |
| 01:41.21 | beters | whats the escape character? |
| 01:42.16 | Katty | echo '${var1} ${var2}' | moo |
| 01:42.30 | beters | exten => 900,n,System(/usr/bin/lynx -dump http://www.xxxx.com/alltest/xxx/index.php?callerid=0&zip=${APCODEE} > output.log) |
| 01:42.44 | beters | breaks at & |
| 01:44.13 | Katty | well |
| 01:44.26 | Katty | you could echo a line using a string like ### |
| 01:44.51 | Katty | and then change foo.txt "###" "&" |
| 01:45.32 | Katty | and then pipe that to lynx |
| 01:45.47 | beters | yeah |
| 01:46.00 | beters | seems like there would be an escape character though |
| 01:46.32 | Katty | \E? |
| 01:46.52 | Katty | or \e |
| 01:47.36 | Katty | dunno if that's doable tho |
| 01:49.45 | beters | ah |
| 01:49.53 | beters | there is an escape |
| 01:50.02 | beters | you have to quote it and slash it |
| 01:50.07 | beters | "\&" |
| 01:50.13 | beters | to make it escape |
| 01:50.25 | Katty | (= |
| 01:50.32 | beters | awesome |
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| 01:56.12 | *** part/#asterisk danryson (n=linda@74-92-154-149-WashingtonDC.hfc.comcastbusiness.net) |
| 01:58.56 | Katty | bored. |
| 01:59.55 | voipmonk | make sushi |
| 02:02.20 | Katty | is watching 9 |
| 02:05.17 | ChannelZ | NEIN! NEIN! |
| 02:20.46 | eppigy | Katty: OMG I JUST SAW THAT LAST NIGHT |
| 02:21.23 | Katty | so far it's kinda cute. |
| 02:21.30 | eppigy | it is pretty good |
| 02:21.35 | eppigy | I like the visuals |
| 02:50.04 | jblack | scratches his head. |
| 02:50.27 | jblack | VA software became sourceforge inc, right? Well, now lnux is "Geeknet, Inc" |
| 02:57.00 | *** join/#asterisk jsmith (n=nnjsmith@asterisk/training-and-documentation-guru/jsmith) |
| 02:57.00 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 02:57.22 | jsmith | jaytee: You around? |
| 02:57.30 | jaytee | yes |
| 02:57.52 | jaytee | what's up? |
| 02:57.57 | jsmith | See private msg, please |
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| 03:14.24 | Katty | eppigy: that was a cute movie. |
| 03:14.26 | Katty | eppigy: kind of an open ending tho :< |
| 03:14.38 | eppigy | yeah I agree |
| 03:20.20 | Katty | still very cute |
| 03:22.54 | jblack | Oh, a movie? |
| 03:23.23 | jblack | Katty: Is it worth seeing? |
| 03:24.06 | Katty | i enjoyed it. |
| 03:24.15 | Katty | i would consider it a children's movie |
| 03:25.19 | jblack | eww yuck. |
| 03:25.40 | jblack | There's some bad critter news tonight |
| 03:26.01 | Katty | oh? |
| 03:26.47 | jblack | huh. Learn somethign every day. Did you know that animal sacrifice is protected/ |
| 03:27.14 | jblack | http://www.cnn.com/2009/CRIME/12/30/pennsylvania.animal.remains/index.html |
| 03:27.42 | jblack | rofl. Captain Picard is now Sir Picard. |
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| 03:55.02 | ModernGeek | a |
| 03:55.08 | ModernGeek | Reason: Q.850;cause=29;text="FACILITY_REJECTED" |
| 03:55.17 | ModernGeek | Can anyone tell me about a 501 error that would give that back? |
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| 03:55.30 | ModernGeek | w/b |
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| 04:13.48 | ChannelZ | ok who knows svn? |
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| 04:34.19 | ecrane | ChannelZ: I know svn. He owes me $5..... |
| 04:34.43 | ecrane | ChannelZ: I know a bit about it... what you wanna know? |
| 04:34.49 | ChannelZ | heh :) |
| 04:34.56 | ChannelZ | I think I got myself going |
| 04:35.06 | bcrisp | 1000 ways to die is awesome |
| 04:38.52 | bcrisp | ...---... |
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| 05:07.24 | sun28 | moin |
| 05:07.41 | ChannelZ | honks |
| 05:07.55 | sun28 | *shrugs* |
| 05:08.36 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-xjjcbkiijjkqgloo) |
| 05:12.10 | ChannelZ | someone told me I should do that if I love jesus |
| 05:19.22 | sun28 | $ drink < bottle; opener |
| 05:19.22 | sun28 | bottle: cannot open |
| 05:19.22 | sun28 | opener: not found |
| 05:19.44 | sun28 | ChannelZ: |
| 05:19.45 | sun28 | <PROTECTED> |
| 05:19.45 | sun28 | ar: God does not exist |
| 05:19.51 | sun28 | :( |
| 05:27.44 | *** join/#asterisk VoIPMasta (n=fabio@201.139.128.244.cable.dyn.cableonline.com.mx) |
| 05:27.52 | Kobaz | i remember fixing a compile bug by fixing a path somewhere |
| 05:27.53 | VoIPMasta | Hi there, anyone experienced with .call files? |
| 05:27.57 | Kobaz | er, wrong window |
| 05:32.56 | sun28 | VoIPMasta: yeah |
| 05:33.57 | VoIPMasta | sun28, I'm trying to have a call file that will dial my cellphone and then another number (previously set), so that I can have long distance calls in my mobile phone. Both calls are placed through an external VoIP provider, I've managed to have my cell phone dialed and then the other number, however I can't "bridge" audio between both calls |
| 05:35.08 | VoIPMasta | sun28, http://pastebin.ca/1732717 |
| 05:35.13 | VoIPMasta | sun28, That's what I have so far |
| 05:38.59 | VoIPMasta | sun28, any ideas on how to make both calls "talk to each other"? |
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| 06:39.43 | kaseano | hi, does anyone know if you can get a 900 voip number that bills per minute without getting a customer's cc |
| 06:40.31 | kaseano | I Think that might be what the PPM thing on a lot of voip sites too, I'm not sure tho |
| 06:55.09 | jblack | I would love a 900 number for a home number. |
| 06:59.46 | ChannelZ | end telemarketing forever |
| 07:04.41 | *** join/#asterisk VxJasonxV (n=jason@xmms2/troll/VxJasonxV) |
| 07:06.13 | VxJasonxV | Two questions: |
| 07:06.41 | VxJasonxV | (1) are there any good docs on res_phoneprov? More specifically with polycom phones? |
| 07:06.57 | VxJasonxV | googling it gives me mailing lists of, and direct commit logs |
| 07:09.01 | VxJasonxV | (2) I'm trying to register with Gizmo as a SIP provider, and I'm getting really weird errors. The main one is "check_auth: username mismatch, have <jason>, digest has <17476199487>". Frankly, I have absolutely no idea why it's mentioning my name, that definition doesn't exist in the peer registration, only in my own extension registration... |
| 07:10.03 | ChannelZ | show us sip.conf |
| 07:11.24 | ecrane | In dialplan, how do I get the phone number of the person who is calling me and assign it to a variable? |
| 07:12.28 | ChannelZ | generally that's the caller ID number - it it's self can be treated like a variable, ${CALLERID(num)} |
| 07:15.04 | ecrane | But if the caller id has something more specific, like a name in it, then I get the name ;<. Anywhere I might be guaranteed to get the phone number? (assuming I received a phone number originally...) |
| 07:15.31 | VxJasonxV | http://pastie.org/private/hvzwnuijha4lkawdy1vig |
| 07:15.33 | ChannelZ | CALLERID(num) should be just the number, CALLERID(name) is the name. |
| 07:15.40 | VxJasonxV | This is Asterisk 1.6.1.12, btw |
| 07:16.11 | ecrane | ChannelZ: ah.. thanks! (duh, oops heh!) |
| 07:16.37 | ChannelZ | Vx: are you getting this error on an incoming call or when it's trying to register? |
| 07:16.45 | *** join/#asterisk sebbl (n=Momofu@cable-195-14-235-105.netcologne.de) |
| 07:17.10 | VxJasonxV | on register |
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| 07:26.59 | ChannelZ | I'm a little confused because usually that's seen on the side the peer is registering to, I think |
| 07:28.38 | ChannelZ | basically it's saying the thing registering is trying to use '17476199487' as an authentication username, while * is expecting 'jason'. |
| 07:29.30 | VxJasonxV | I don't understand why * is expecting my name |
| 07:29.41 | ChannelZ | what does 'sip show registry' say |
| 07:30.27 | VxJasonxV | no registrations |
| 07:30.37 | ChannelZ | uhh |
| 07:31.00 | ChannelZ | are you sure you're editing the right file/have reloaded sip? |
| 07:32.43 | ChannelZ | was that your complete sip.conf or did you just cut out a section? |
| 07:33.36 | *** join/#asterisk tzafrir__laptop (n=tzafrir@local.xorcom.com) |
| 07:34.24 | VxJasonxV | well, that was just the registration for gizmo. I the only other thing I have in there is my user extension registration. |
| 07:34.32 | ChannelZ | cuz if that was your whole thing, it's probably not parsing right and even doing anything |
| 07:35.03 | VxJasonxV | err, it's definitely missing the globals |
| 07:35.10 | VxJasonxV | I'm assuming you'd like me to repaste it |
| 07:35.25 | ChannelZ | if I made a register => the first line in sip.conf the whole thing busts.. I think you need a [general] section, even if empty. |
| 07:37.25 | VxJasonxV | I have a general section |
| 07:37.31 | VxJasonxV | it was indeed a very small clip |
| 07:38.10 | ChannelZ | ok.. but 'sip show registry' literally lists nothing? |
| 07:38.44 | VxJasonxV | it has the column header, but only says "0 SIP registrations" |
| 07:38.54 | VxJasonxV | http://pastie.org/private/n9hgonsp1flvi97py3g is a "corrected" paste |
| 07:39.08 | VxJasonxV | I'll note that my user has more entries, much more |
| 07:39.25 | VxJasonxV | I just didn't feel they were pertinent. |
| 07:39.55 | ChannelZ | take your register line out of that [authentication] section, it should be in [general] |
| 07:40.23 | VxJasonxV | ah, oops |
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| 07:40.43 | ChannelZ | the message you were getting was Gizmo trying to contact YOU, not the other way around |
| 07:41.23 | VxJasonxV | ah, heh |
| 07:41.25 | ChannelZ | (or maybe not even gizmo actually, I have no idea as you didn't paste the whole thing) |
| 07:42.35 | VxJasonxV | MUCH better |
| 07:45.19 | ChannelZ | as to your first question all I can offer is http://www.voip-info.org/wiki/view/snom+mass+deployment |
| 07:45.42 | VxJasonxV | yeah, I've seen that |
| 07:47.11 | ChannelZ | shrugs - never used res_phoneprov |
| 07:47.27 | ChannelZ | My phones are linksys and I provision them separately with tftp |
| 07:48.07 | VxJasonxV | what that page doesn't answer is, how to configure much of the asterisk end. phoneprov status says the http server isn't running, even though I'm pretty sure I have the file properly configured and, well, there shouldn't really be a start/stop, it should just run |
| 07:48.19 | VxJasonxV | but, I'll work on that tomorrow. |
| 07:48.38 | VxJasonxV | The dialplan still confuses me. |
| 07:48.50 | VxJasonxV | When it comes to gizmo users calling me... it'll use the default dialplan, right? |
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| 07:55.42 | ModernGeek | is there a log of the room anywhere? |
| 07:55.49 | ModernGeek | I asked a question earlier, and I think I saw an answer. |
| 07:55.54 | ModernGeek | and accidently closed my client |
| 07:59.51 | ChannelZ | What was your question? I have some scrollback |
| 08:01.11 | ChannelZ | Vx: Normally it goes into the default *context*, yes |
| 08:02.15 | ChannelZ | But you can read more in the default sip.conf - if you want to control where it goes beyond just the default context, you need to setup a sip peer for the provider |
| 08:04.58 | ecrane | core |
| 08:05.56 | ecrane | ChannelZ: thanks again for the help. Got everything working now. |
| 08:06.19 | ChannelZ | sure |
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| 09:34.21 | Jenna | Hiyall. I seem to have a tricky situation. I have TDM400P with 2 FXO ports. (channel 2,3) I have a PSTN line comming in on channel 2 (incoming out going calls work fine on this), while I have a plain old phone connected to channel 3. I cant seemt to call that phone thru asterisk ( eg. exten => 123,1,Dial (Zap/3)) any ideas why ? |
| 09:35.32 | Tim_Toady | you need an FSX to connect a phone not an fxo |
| 09:35.46 | Jenna | this channel 3 (zap/3) has also got signalling=fxs_ks configured in zapata-channels.conf |
| 09:36.42 | Jenna | yeah, that did cross my mind. but I am able to call out from my FXO port 2 (channel 2) |
| 09:36.49 | Tim_Toady | fxsks is for fxo ports you need an fxs module and you can connect ur phone there |
| 09:37.26 | Tim_Toady | http://www.voip-info.org/wiki/view/FXO |
| 09:37.38 | Jenna | hmm. so your saying that there is no way I can make my FXO port mimic FXS port |
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| 09:37.43 | Tim_Toady | nope |
| 09:38.16 | Jenna | not even by channing the signalling to fxo_ks |
| 09:38.36 | Tim_Toady | no, you need an fxs module for ur card |
| 09:38.52 | Jenna | okay. thanx for clearing that up |
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| 10:07.47 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 10:11.26 | lesouvage | I have Dial(SIP/508&SIP/504&SIP/505&SIP/506&SIP/trunk-paris-lyon/openbts,40,rt) in my dialplan. Only the phones of "Lyon" ring. When I remove SIP/trunk-paris-lyon/openbts the "Paris" phones ring. Is there an explanation for this unexpected behaviour? |
| 10:15.39 | [TK]D-Fender | lesouvage: Go look at the actual call |
| 10:18.54 | ChannelZ | yeah I wonder if the remote trunk-paris-lyon side is actually answering the call and then ringing on it's own end |
| 10:19.16 | tzafrir__laptop | btw: you use openbts? (just wondering) |
| 10:20.20 | lesouvage | [TK]D-Fender: I'm testing with a local sip call with Zoiper. If I dial local channels and add a delay of 5 seconds in using the trunk (or whatever you call it) 508, 504, 505 and 506 stop ringing as soon as Lyon starts to ring. |
| 10:20.26 | ChannelZ | hmm.. so I search for 'alarm clock' on amazon and it's showing me an ottoman |
| 10:21.53 | lesouvage | ChannelZ: I think that i it. I think I add an Answer() on the wrong place. Thanks, I look into it but I think that is the solution. |
| 10:23.38 | lesouvage | tzafrir__laptop: Yes, I started my R&T OpenBTS project and it is really amazing. It is not perfect but is working quite well and my opinion is that OpenBTS has great potentials and will be really disruptive. |
| 10:23.55 | lesouvage | R&T= research and try |
| 10:30.00 | lesouvage | ChannelZ: It is working now. Was trying to fix it on the wrong Asterisk box :-( Thanks ! |
| 10:30.41 | ChannelZ | sure |
| 10:34.49 | m0e | what's a good fax tutorial? (for incoming) |
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| 10:44.18 | ChannelZ | dunno I have a machine |
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| 11:07.41 | zeno | hi all |
| 11:08.10 | zeno | guys can you help me with a problem on a FXO port? |
| 11:09.58 | TSM | mabey mabey not |
| 11:10.17 | zeno | i have 4 PSTN lines. 3 of them are configured on 3 FXO ports, but the fourth has a problem: the line comes from the Telco and in the office is filtered... the filter is used because without it we listen other people.. there is a lot of noise |
| 11:10.39 | zeno | the problem is the forth line is not managed by asterisk |
| 11:10.44 | zeno | *fourth |
| 11:11.22 | zeno | when i make a call to this line i hear the ringing tone, but asterisk doesn't manage the incoming ring |
| 11:11.39 | zeno | the same if i try to call from that line to the outside world |
| 11:11.47 | zeno | atsreisk doesn't manage it |
| 11:11.54 | zeno | if i remove the filter everything is ok |
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| 11:12.15 | zeno | but, yiou know, i need the filter to reduce the noise |
| 11:14.18 | zeno | i registered the dialtone of the line with the filter and the frequency is the same of others... i am desperate! |
| 11:31.11 | *** part/#asterisk Tech_Travis (n=Administ@cpe-76-87-9-130.socal.res.rr.com) |
| 11:35.45 | soman | Hi, I am running asterisk 1.6 with TE121 T1/E1 card. while trying to record the conference, i am getting the error.. please check this http://pastebin.com/m504db656. can anyone help what could be the problem |
| 11:36.49 | soman | ChannelZ: can you help me what could be the problem |
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| 11:41.11 | TSM | zeno: what is this "filter", have you got a split PSTN line using a spliter |
| 11:41.33 | kaseano | hey, anyone know if there are #900 voip lines ? |
| 11:41.49 | kaseano | or a PPM line that doesn't require a cc, just bills the phone number |
| 11:41.53 | TSM | soman: have you configured your TE121 card yet? dahdi_genconf i think |
| 11:42.26 | TSM | soman: did you start dahdi before asterisk |
| 11:42.26 | soman | TSM: yes.. i ran dahdi_genconf |
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| 11:42.52 | soman | TSM: i stopped asterisk, started dahdi.. and then started asterisk |
| 11:42.53 | benngard | cold in gothenburg/sweden |
| 11:42.55 | m0e | soman, "dahdi show channels" you sure its configured correctly? |
| 11:44.03 | TSM | what does "service dahdi status" show? |
| 11:45.35 | soman | TSM: dahdi status shows "### Span 1: WCT1/0 "Wildcard TE121 Card 0" (MASTER)" and lists 1-24 with PRI |
| 11:46.09 | kaseano | anyone know if you can get a voip 900 number, or if I have to gateway to a pots 900 |
| 11:47.22 | TSM | soman: do in asterisk CLI "pri show spans" and "dahdi show channels", this should tell you if asterisk knows of the pri |
| 11:48.55 | soman | TSM: in asterisk CLI, when i run the above commands.. it says "No such command" |
| 11:49.37 | m0e | soman, dahdi show channels gives a "No such command"? |
| 11:50.08 | soman | m0e: yes, it says no such command try help |
| 11:50.13 | Jenna | I've installed asterisk from debian repo, I wish to know if it has been compiled with "ODBC storage for Voicemail". Is there an easy way to find out. |
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| 11:51.37 | m0e | soman, are you using centos/asterisk repo? If so, I think you dont have everything installed |
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| 11:52.08 | soman | TSM:m0e: i am ruuning asterisk on Cent OS release 5.4 |
| 11:52.40 | soman | I have not installed asterisk from centos repo... i have installed manually |
| 11:53.09 | tzafrir__laptop | soman, try: module unload chan_dahdi.so |
| 11:53.16 | tzafrir__laptop | module load chan_dahdi.so |
| 11:53.35 | tzafrir__laptop | do you get any errors? Do you get a 'dahdi' command completed by tab? |
| 11:54.39 | soman | tzafrir__laptop: i tried.... it unloaded the module, but while loading gave same error |
| 11:54.52 | soman | ERROR[1802]: chan_dahdi.c:8741 mkintf: Unable to open channel 1: No such device or address |
| 11:54.52 | soman | here = 0, tmp->channel = 1, channel = 1 |
| 11:55.01 | soman | <PROTECTED> |
| 11:55.44 | tzafrir__laptop | soman, please pastebin the output of: dahdi_hardware; lsdadhi |
| 11:55.48 | TSM | you dahdi config may be wrong dont knw |
| 11:57.15 | soman | here is dahdi_hardware and dahdi_scan http://pastebin.com/m5eaa17c5 |
| 11:57.48 | TSM | soman: where are you? |
| 11:58.52 | TSM | also PB your /etc/dahdi/system.conf |
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| 11:59.05 | TSM | soman: in india? |
| 11:59.06 | tzafrir__laptop | soman, I asked for lsdahdi, not dahdi_scan |
| 12:00.02 | soman | here is my chan_dahdi.conf and /etc/dahdi/system.conf http://pastebin.com/m5b37cd5e |
| 12:00.54 | TSM | well its not syncing up, the config in your system.conf does not match up to dahdi_scan |
| 12:01.00 | soman | here is lsdahdi output http://pastebin.com/m5b6fe504 |
| 12:02.12 | soman | TSM: can you tell me what needs to be corrected |
| 12:02.19 | tzafrir__laptop | soman, several issues: |
| 12:02.32 | TSM | loadzone * defaultzone, i have them at the top |
| 12:02.44 | tzafrir__laptop | 1. dahdi_dummy is loaded. It causes all span numbers to have an offset |
| 12:03.06 | TSM | dahdi_dummy only loads because its not loaded a proper card |
| 12:03.21 | tzafrir__laptop | 2. you have only one span now, but one extra span in the configuration. What's the output of: dahdi_hardware -v |
| 12:03.28 | TSM | span=1,0,0,ccs,hdb3,crc4 |
| 12:03.28 | TSM | bchan=1-15,17-31 |
| 12:03.28 | TSM | hardhdlc=16 |
| 12:03.38 | TSM | this is my system.conf, UK E1 |
| 12:04.11 | TSM | i have hardhdlc as i have a sangoma card im not sure if the digium cards are the same |
| 12:04.36 | TSM | span is 1,0,0 as the remote end of the line is ususaly MASTER timing |
| 12:04.45 | TSM | thats the second digit |
| 12:05.43 | soman | pci:0000:08:08.0 wcte12xp+ d161:8000 Wildcard TE121 is the output for dahdi_hardware -v |
| 12:05.48 | TSM | and your chan_dahdi.conf i think is the wrong way around |
| 12:06.22 | TSM | switchtype=euroisdn |
| 12:06.22 | TSM | context=from-pstn |
| 12:06.22 | TSM | group=0 |
| 12:06.22 | TSM | echocancel=no |
| 12:06.22 | TSM | signalling=pri_cpe |
| 12:06.22 | TSM | channel =>1-15,17-31 |
| 12:06.28 | TSM | bah i cant be arsed with PB today there is hardly anyone using the room |
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| 12:20.18 | TSM | mabey you have to change my hardhdlc to dchan |
| 12:20.18 | TSM | and you may have to put.... |
| 12:20.18 | TSM | echocanceller=mg2,1-15,17-31 |
| 12:20.18 | *** join/#asterisk mahlon (i=mahlon@martini.nu) |
| 12:20.18 | soman | ok.. let me try with your config |
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| 12:20.19 | tzafrir__laptop | soman, hmm... sorry, confused your output with an older output. Ignore what I wrote about dahdi_dummy and such |
| 12:20.19 | tzafrir__laptop | The problem is that you didn't run dahdi_cfg |
| 12:20.19 | tzafrir__laptop | (which is run automatically anyway as part of /etc/init.d/dahdi start |
| 12:20.19 | tzafrir__laptop | soman, the config you had is for a T1 card. Your card is T1 |
| 12:20.19 | soman | yep |
| 12:20.19 | tzafrir__laptop | If you want to make it E1, you should first make it E1 (either through a jumper or through a module param - e1t1override, IIRC) |
| 12:20.20 | soman | TSM: even with your config i am getting the same error when i try dahdi start |
| 12:20.20 | tzafrir__laptop | if you fail to do that, no use in configuring it as E1 |
| 12:20.20 | *** join/#asterisk carrar (i=tim@osburn.com) |
| 12:20.20 | tzafrir__laptop | dahdi_genconf would configure it as E1 when it will see an E1 there |
| 12:20.20 | *** join/#asterisk hohum (i=dcorbe@apollo.corbe.net) |
| 12:20.20 | soman | tzafrir__laptop: that might be the reason, i am trying to configure it as E1, the problem is iam in india.. and the asterisk server is in finland |
| 12:20.20 | soman | can you tell me how to configure that as E1 from remote |
| 12:20.21 | TSM | you cant |
| 12:20.21 | TSM | just read the TE121 has a jumper as tzafrir |
| 12:20.21 | TSM | said |
| 12:20.21 | tzafrir__laptop | setting a jumper is remotely possible to do remotely |
| 12:20.21 | soman | TSM: yep. only then will i be able to configure it properly? |
| 12:20.22 | tzafrir__laptop | I'd go for the t1e1override option |
| 12:20.29 | soman | tzafrir__laptop: ;) good one |
| 12:20.53 | soman | how to override that setting |
| 12:20.53 | TSM | duh, why do they bother adding a jumper, dumb, i use sangoma, none of that shite |
| 12:20.53 | kaseano | anyone know if there are 900 voip lines, or do I have to get a pots and gateway it |
| 12:21.05 | TSM | kaseano: what? |
| 12:21.12 | tzafrir__laptop | basically add a line such as 'option wcte12xp t1e1override=1' to a file under /etc/modprobe.d |
| 12:21.32 | TSM | or mod the one that is there |
| 12:21.41 | tzafrir__laptop | then unload all modules and load them |
| 12:21.50 | kaseano | TSM, can you buy/rent a voip #900 line, that charges the customer like a 900 line typically does |
| 12:21.58 | tzafrir__laptop | though before you do that, consider: dahdi_genconf modules |
| 12:22.09 | tzafrir__laptop | will save you some time on that restart |
| 12:22.28 | soman | shall i add that line to /etc/modprobe.d/dahdi file |
| 12:22.43 | tzafrir__laptop | and then: /etc/init.d/dahdi restart |
| 12:22.50 | tzafrir__laptop | dahdi_genconf |
| 12:22.53 | tzafrir__laptop | dahdi_cfg |
| 12:23.21 | kaseano | stupidly I bought a bunch of 800 POTS lines and used an adapter to convert them to voip to send to asterisk |
| 12:23.35 | kaseano | making sure I don't make the same mistake w #900 lines |
| 12:24.21 | TSM | kaseano: what you talking about 800 POTS???? you US based and talking about free and premium rate? |
| 12:25.26 | soman | ok.... will catch you later |
| 12:25.36 | soman | thanks for the support |
| 12:25.43 | soman | wish you a very happy new year... |
| 12:25.58 | kaseano | TSM yeah i'm talking about free & premium rates. |
| 12:26.03 | kaseano | lol |
| 12:27.51 | TSM | kaseano: from what ive read its not possable but i dont know, have you thought of just getting it all supplied via a PRI? |
| 12:28.06 | kaseano | I'm not sure what a PRI is, let me google it real quick =) |
| 12:28.24 | TSM | PRI/T1/E1 |
| 12:28.28 | TSM | full digital line |
| 12:28.33 | TSM | instead of POTS |
| 12:28.40 | kaseano | oh right |
| 12:28.44 | kaseano | well |
| 12:28.55 | kaseano | I'm trying to bill via the phone line and avoid getting credit card info |
| 12:28.57 | fenrus | Why not go with SIP ? |
| 12:29.07 | kaseano | only way to do that seems to be w a 900 POTS |
| 12:29.18 | TSM | naa there must be 900 via PRI |
| 12:29.19 | kaseano | except I run everything through asterisk, which means I'd have to adapt the pots |
| 12:29.27 | kaseano | that's what I'm really hoping =) |
| 12:29.31 | kaseano | wondering if anyone here knew |
| 12:29.33 | TSM | if you get pri then its much easier to interface to asterisk |
| 12:30.16 | kaseano | I couldn't find anything on google so I wasn't sure if they even existed |
| 12:30.33 | TSM | what?? 900 or PRI? |
| 12:30.50 | kaseano | a 900 digital line |
| 12:31.10 | TSM | usualy anything that is provided over POTS can be put on PRI |
| 12:31.17 | TSM | usualy in us its refered to as T1 |
| 12:31.26 | TSM | its not as good as a E1, but hey |
| 12:31.33 | TSM | you cant get E1 |
| 12:31.38 | kaseano | what's E1? |
| 12:31.49 | TSM | E1 is the Euro standard of T1 |
| 12:31.57 | kaseano | oh ok, so it wouldn't be international |
| 12:32.04 | TSM | it has 30 channels while T1 has 24 i think |
| 12:32.16 | kaseano | oh. |
| 12:32.20 | TSM | think of it as a 1Mb vs 2Mb line |
| 12:32.27 | TSM | E1 is better than T1 |
| 12:32.38 | TSM | ultimatly they are the same for sakes |
| 12:33.07 | kaseano | well I haven't bought the 800 voip lines yet either |
| 12:33.15 | kaseano | I was thinking about using vitelity |
| 12:33.25 | kaseano | i'm hoping they'll do something w 900's |
| 12:33.30 | kaseano | I duno if they use an E1 |
| 12:33.44 | TSM | good thing with a T1/E1/PRI is that you can have say 500 numbers on a single line but only 24 concurrent calls etc.. but then you can add extra lines and balance the numbers across all of them |
| 12:34.10 | kaseano | oh I wasn't going to set up the digital numbers myself, just buy/rent them |
| 12:34.38 | TSM | yes but you need to know what number people are calling and lots of lines, you will want digital T1 lines |
| 12:34.41 | kaseano | and send them to my server w asterisk |
| 12:34.50 | fenrus | The good thing with sip is that you dont need to buy expensive timeslots from a telco.. :) |
| 12:35.10 | Jenna | Im not using ael-* & dundi-* plus few other pre-define dialplans. only my create ones i.e (myincoming, myoutboud, myphone etc.). Is it safe for me delete other pre-define dialplans ? |
| 12:35.35 | kaseano | fenrus what's SIP |
| 12:35.42 | TSM | all adds up in the end ive worked out, 24 SIP trunks vs a PRI is nearly the same, cheeper but still |
| 12:36.00 | fenrus | kaseano, Sip is telephony over ip.. |
| 12:36.01 | tzafrir__laptop | infobot, sip |
| 12:36.02 | infobot | somebody said sip was Session Initiation Protocol, http://www.cs.columbia.edu/sip/ (see RFC 3261) |
| 12:36.09 | fenrus | :) |
| 12:36.09 | kaseano | thought that was voip |
| 12:36.10 | tzafrir__laptop | ~wiki sip |
| 12:36.51 | Jenna | and I thought I was a newbie |
| 12:37.19 | tzafrir__laptop | kaseano, if you don't know what $X is, try searching first. e.g. asking infobot in a private message |
| 12:37.23 | fenrus | Jenna, i'd backup the dialplans you think is not in use and try reloading without them |
| 12:37.52 | Jenna | fenrus, good idea |
| 12:38.34 | Jenna | lemme take a backup of extensions.conf & run a slashed-down version |
| 12:39.39 | Jenna | fenrus, what about extension.ael. can I just disable its modules, since Im relying on plain old dialplan style ? |
| 12:40.21 | fenrus | Jenna, i'd say do one thing at the time, easeier to fault-trace if something goes bad.. |
| 12:40.42 | Jenna | yeah. |
| 12:41.28 | Jenna | but it was a different question |
| 12:46.22 | snadge | happy new years and shit |
| 12:53.32 | kaseano | ok so it looks like 900 digital lines don't exist for billing reasons, which I think means I have to adapt the pots number to get it into asterisk right? |
| 12:54.11 | TSM | they do |
| 12:54.13 | TSM | duh |
| 12:54.32 | fenrus | "900 lines", as in 900 continious calls ? |
| 12:54.33 | kaseano | oh awesome =) |
| 12:54.48 | *** join/#asterisk shamelessn00b (n=chatzill@58-65-172-114.nayatel.pk) |
| 12:54.49 | TSM | digital lines are not SIP, digital lines just referres to PRI/T1/E1 or SIP |
| 12:54.49 | fenrus | or as a prefix in the us, +1-900-X ? |
| 12:54.53 | shamelessn00b | Hi guys |
| 12:54.54 | kaseano | yeah the premium rate billed 900 lines in a digital form |
| 12:54.55 | TSM | SIP is via IP |
| 12:54.58 | kaseano | oh |
| 12:55.02 | TSM | T1/E1/PRI is TDM |
| 12:55.11 | kaseano | so they Are SIP |
| 12:55.14 | TSM | its like POTS just digital |
| 12:55.21 | kaseano | OH |
| 12:55.24 | TSM | no SIP is not T1 |
| 12:55.28 | kaseano | haha Wow this is confusing |
| 12:55.39 | shamelessn00b | SIP is RTP |
| 12:55.46 | shamelessn00b | :P |
| 12:56.05 | kaseano | lol |
| 12:56.14 | fenrus | well, it's not that confusing, T1/E1 and PRI are physical cables that your telco provide, you run SIP over an internet-connection |
| 12:56.30 | shamelessn00b | or a LAN |
| 12:56.31 | kaseano | well most importantly, are there 900 numbers that I can direct to my server to talk to asterisk |
| 12:56.36 | fenrus | shamelessn00b, sure thing ;) |
| 12:56.52 | shamelessn00b | hey anyone have experiance on wanrouter |
| 12:56.55 | TSM | yes 900 is just a number, you can have them supplied over PRI/T1 |
| 12:57.05 | TSM | its better allocation of resources |
| 12:57.07 | fenrus | shamelessn00b, what do you mean with "wanrouter" ? |
| 12:57.11 | kaseano | ok. Most 900 providers will probably do that right? |
| 12:57.16 | shamelessn00b | the utility |
| 12:57.19 | TSM | fenrus: sangoma i think |
| 12:57.20 | shamelessn00b | that comes with wanpipe |
| 12:57.30 | shamelessn00b | yeah, its sangoma |
| 12:57.35 | TSM | shamelessn00b: what you want to know? |
| 12:57.40 | Akiraa | Can you get SIP working for a server behind NAT with a dynamically allocated public IP (IP changes upon router reset); also, clients may have similar setups? |
| 12:58.14 | shamelessn00b | the wanpipemon utility can show you how many times your PRI went down, it doesnt show the exact time whenit went down |
| 12:58.15 | Akiraa | without having to force a client use his computer for tunneling etc |
| 12:58.16 | fenrus | Akiraa, yes. |
| 12:58.20 | kaseano | TSM I had one other thing, most of my employees are using cell phones |
| 12:58.27 | shamelessn00b | I wanna see from the logs when the PRIs went down |
| 12:58.42 | kaseano | and will either be receiving calls from, or calling into my box with asterisk |
| 12:58.58 | fenrus | i would not say it's a perfect condition, but it could be done with dyndns and stuff like that.. |
| 12:59.00 | TSM | shamelessn00b: look in your asterisk full log, it usualy will be shown |
| 12:59.04 | kaseano | cell phones Are pots lines, which means I Will need a pots/voip adapter right |
| 12:59.18 | shamelessn00b | var/log/messages?/ |
| 12:59.30 | TSM | var/log/asterisk/full |
| 12:59.40 | TSM | kaseano: you mean you want a GSM gateway? |
| 12:59.45 | fenrus | TSM, shamelessn00b: confusin gwith wanrouter and wanpipe.., i thaught it was some ip-routing related issue.. (: |
| 12:59.49 | TSM | kaseano: you can get IP ones |
| 13:00.14 | kaseano | is that hardware or a service I have to buy |
| 13:00.23 | TSM | kaseano: hardware |
| 13:00.27 | shamelessn00b | fenrus: you can use wanpipemon to TAP your TDM channels and stream them as RTP packets |
| 13:00.28 | shamelessn00b | :P |
| 13:00.40 | fenrus | shamelessn00b, neat. :) |
| 13:00.59 | kaseano | ok |
| 13:01.00 | kaseano | thx TSM =) |
| 13:01.17 | TSM | kaseano: GSM gateways are not cheep but will allow you to have a cellphone connected to you asterisk for cheeper calls |
| 13:01.23 | shamelessn00b | TSM: do you know where does winepipemon read from, I mean which log file does it read from?? |
| 13:01.32 | Akiraa | kaseano: you would need a GSM gateway (can be expensive) and an ATA to connect the gateway to the IPPBX; or you can just get a GSM-VoIP gateway (but that may be more expensive than the pots gateway+ATA way) |
| 13:01.40 | TSM | kaseano: also you have to work out how many channels you want and need that many contracts etc.. |
| 13:02.01 | kaseano | well what's complicated, my server is on the other side of the country |
| 13:02.05 | TSM | Akiraa: just get IP gateways as they are more reliable and call setup time is quicker |
| 13:02.17 | kaseano | so I'd need the hardware at some physical location with at least a T1 right |
| 13:02.23 | TSM | Akiraa: they are not too bad in cost, look at portec i think |
| 13:02.43 | TSM | kaseano: yes, T1 is usualy available everywhere as it goes long distance |
| 13:03.19 | Akiraa | fenrus: So how do you handle SIP in the server/client multi NAT configuration? |
| 13:03.20 | kaseano | do you know if you can hire a company to just convert gsm - voip ? |
| 13:03.35 | TSM | kaseano: good thing with IP GSM gateways is they dont have to be in the same location as the asterisk server, you can have them dotted around the country for better tarrifs etc if you wanted |
| 13:03.48 | kaseano | oh awesome ! |
| 13:04.11 | TSM | kaseano: even diffrent countries if you wanted |
| 13:04.47 | Akiraa | kaseano: would involve either a custom PCB (very expensive), or some custom FPGA + PCB (still expensive) |
| 13:05.04 | fenrus | Akiraa, STUN might be what you want.. |
| 13:05.58 | fenrus | Akiraa, this of course requires some port-forwarding in the nat-router in front of the pbx |
| 13:06.09 | kaseano | googling PCB =) |
| 13:07.20 | Akiraa | kaseano: design, programming, testing of custom PCBs is only worth it if you want to become a producer, really :) |
| 13:07.42 | TSM | Akiraa: why custom PCB???? |
| 13:08.01 | kaseano | no. Just trying to find the quickest cheapest way to convert pots - voip |
| 13:10.28 | TSM | kaseano: just get them on PRI, its cheeper in the longer run, look at the cost of 24 ports of POTS->VOIP vs a single PRI |
| 13:11.19 | kaseano | yeah you're probably right. |
| 13:11.30 | kaseano | I have some developers working from home and moving around a lot tho |
| 13:12.00 | kaseano | so they won't always have a great internet connection |
| 13:12.00 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 13:12.00 | kaseano | which means the quality of the PRI will be heavily degraded right ? |
| 13:12.19 | TSM | no, PRI is digital, it connects directly to your asterisk server |
| 13:12.31 | TSM | no quality loss as there is no conversion |
| 13:12.41 | kaseano | well the employees are on cell phones tho |
| 13:12.52 | TSM | and where does that come into it |
| 13:12.57 | kaseano | you meant convert them to pri right ? |
| 13:13.06 | TSM | no i mean your POTS lines |
| 13:13.11 | kaseano | well they get routed customer service calls through asterisk |
| 13:13.23 | kaseano | cell phones are POTS aren't they ? |
| 13:13.26 | TSM | speak to your telco |
| 13:13.32 | TSM | this is not hard |
| 13:13.40 | kaseano | ok =) |
| 13:13.46 | TSM | forget about your cell phones |
| 13:13.55 | TSM | get your incomming calls sorted first |
| 13:14.04 | TSM | then look at how you route them out |
| 13:14.21 | TSM | mabey you can just send the call back out over the PRI to them |
| 13:15.16 | kaseano | ok ! |
| 13:17.33 | shamelessn00b | TSM: any idea where does wanpipemon read from (PRI status history) |
| 13:20.58 | tzafrir__laptop | kaseano, there are existing gsm <-> chips. but what do you need to convert to what? |
| 13:21.08 | tzafrir__laptop | how many concurrent calls? |
| 13:21.18 | kaseano | actually just got help from someone and |
| 13:21.36 | kaseano | I have an 800 number that ppl are dialing into |
| 13:21.39 | kaseano | can asterisk just |
| 13:21.47 | kaseano | dial a cell phone with that 800 number ? |
| 13:21.56 | kaseano | or from a different number, to convert pots to voip ? |
| 13:22.39 | Akiraa | In the sip_nat.conf file there is an entry called "externip" -- since my IP is dynamic, is it possible to specify a domain name instead? i.e. xxx.dyndns.org |
| 13:22.51 | kaseano | well ** asterisk calls a pots cell phone with a voip number*** |
| 13:22.54 | kaseano | are cell phones even pots ? |
| 13:25.49 | *** join/#asterisk RypPn (i=TuMbL@rosscom.co.uk) |
| 13:27.25 | voipmonk | Akiraa: go for it |
| 13:31.01 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:07.32 | *** join/#asterisk Godfather_ (n=Godfathe@62.43.134.46.dyn.user.ono.com) |
| 14:10.33 | Katty | it's the last day of 2009 |
| 14:10.39 | Katty | and it will be gone, forever |
| 14:10.57 | Kobaz | foooorrreeeevvveeeer |
| 14:14.02 | TSM | so what |
| 14:14.38 | Katty | so WHAT are you going to do with the very last day of this year? |
| 14:15.33 | zeno | TSM, you there? You said: what is this "filter", have you got a split PSTN line using a spliter --> Nope.. it's a strange filter to reduce noise. I've never seen it before... The line comes into the filter and exit from the filter. If i connect my FXO to the line before the filter i hear a lot of voices (people who are calling maybe on others PSTN line out of my office), but asterisk works fine... if i connect the line |
| 14:15.34 | zeno | after the filter asterisk doesn't work: it doesn't "hear" the in coming call or the dialed numbers inside the office! |
| 14:16.23 | TSM | zeno: it must be messing up with line voltage and impedance which may be why the FXO card does not work correctly |
| 14:16.28 | TSM | anyways i need to go home |
| 14:16.46 | TSM | just get the telco to sort the line |
| 14:16.46 | zeno | TSM, do you know how can i verify? |
| 14:16.51 | TSM | not bodge it |
| 14:17.00 | zeno | ah ok... thanks a lot! |
| 14:17.07 | zeno | so i need to call the telco :) |
| 14:17.20 | zeno | thanks a lot TSM |
| 14:19.04 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:21.37 | *** join/#asterisk alrs (n=lars@98.sub-75-214-211.myvzw.com) |
| 14:21.54 | *** join/#asterisk neurosys (n=neurosys@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 14:22.18 | Katty | anyone have a new year's resolution yet? |
| 14:22.42 | voipmonk | I will call more people |
| 14:23.16 | fenrus | =) |
| 14:24.23 | Katty | call more people? ^_- |
| 14:24.23 | Katty | more...family? |
| 14:24.37 | voipmonk | no |
| 14:24.41 | Pan3D | I will call more random people |
| 14:24.47 | Pan3D | it's funner that way |
| 14:24.47 | Katty | for what porpoise |
| 14:24.59 | fenrus | for fun? |
| 14:25.07 | voipmonk | to help them |
| 14:25.13 | Katty | voipmonk: to help them do what? |
| 14:25.21 | voipmonk | whatever they like |
| 14:25.42 | Katty | so if i call you, then you'll help me clean up my house? :P |
| 14:25.52 | voipmonk | I can call someone to help you do that |
| 14:25.58 | voipmonk | :) |
| 14:27.58 | Katty | but i don't want someone random cleaning my house |
| 14:28.00 | Katty | i want you mister monk! |
| 14:29.58 | *** join/#asterisk hippiehacker (n=user@71.20.113.18) |
| 14:32.27 | shamelessn00b | originate DAHDi/3/03025123456 application Playback(/root/main_menu.wav) |
| 14:32.29 | *** join/#asterisk vally (i=vally@ip-95-222-216-119.unitymediagroup.de) |
| 14:32.32 | shamelessn00b | is this the correct syntax?? |
| 14:32.39 | shamelessn00b | originate DAHDi/3/03025123456 application Playback(/root/main_menu.wav) |
| 14:32.42 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 14:33.06 | shamelessn00b | originate DAHDI/3/03025123456 application Playback(/root/main_menu.wav)* |
| 14:33.22 | *** join/#asterisk Akiraa (n=Akiraa@79.112.38.145) |
| 14:36.04 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 14:42.51 | hippiehacker | having some trouble with a d-channel on a TE110P zaptel |
| 14:43.24 | hippiehacker | http://paste.lisp.org/display/92861 #pastbin of my troubleshooting |
| 14:46.14 | jsmith | hippiehacker: What does your zapata.conf file look like? |
| 14:46.40 | jsmith | hippiehacker: Did you set a switchtype and also set your signalling to either pri_net or pri_cpe? |
| 14:49.10 | *** join/#asterisk shamelessn00b (n=chatzill@58-65-172-114.nayatel.pk) |
| 14:49.15 | shamelessn00b | [Dec 31 19:47:36] WARNING[9815]: channel.c:3106 set_format: Unable to find a cod ec translation path from 0x4 (ulaw) to 0x40 (slin) :( |
| 14:49.34 | shamelessn00b | SIP to SIP call, I'm facing this prob |
| 14:49.45 | shamelessn00b | if I make dahdi call file plays back normally |
| 14:49.49 | jaytee | internal SIP to SIP? |
| 14:49.54 | shamelessn00b | yes |
| 14:50.16 | shamelessn00b | SIP call to an extention defined in dialplan |
| 14:50.16 | jaytee | make sure both phones allow=ulaw and deny everything else |
| 14:50.19 | alrs | shamelessn00b: why are you using a weirdo codec like slin? |
| 14:50.25 | Katty | ryan's truck is still parked outside |
| 14:50.25 | shamelessn00b | Im not |
| 14:50.26 | Katty | and it's 9am |
| 14:50.32 | jsmith | shamelessn00b: Can you pastebin the output of "core show translation"? |
| 14:50.55 | jsmith | alrs: slin = signed linear, and Asterisk uses slin to transcode to/from most other codecs |
| 14:51.00 | ManxPower-work | sln is the itermediat raw "codec" used when doing transcoding. |
| 14:51.48 | shamelessn00b | http://pastebin.com/m5b445d02 |
| 14:52.54 | hippiehacker | switchtype = national and signalling = pri_cpe |
| 14:52.55 | hippiehacker | <PROTECTED> |
| 14:53.04 | *** join/#asterisk Hikah (n=jfoote@83-244-221-226.cust-83.exponential-e.net) |
| 14:53.22 | hippiehacker | updated my pastbin |
| 14:54.42 | *** join/#asterisk Akiraa (n=Akiraa@79.112.38.145) |
| 14:54.52 | hippiehacker | jsmith: we had a power spike/outage, not sure if the card is fried |
| 14:54.56 | shamelessn00b | jsmith: any idea?? |
| 14:55.14 | jsmith | is looking at both pastebins... back in a minute! |
| 14:55.42 | jsmith | shamelessn00b: Uh, yeah... looks like you haven't loaded any of the codec_*.so modules |
| 14:56.11 | shamelessn00b | I have a default installation of asterisk used safe_asterisk to start the server |
| 14:56.16 | shamelessn00b | what am I doing wrong?? |
| 14:57.14 | jsmith | shamelessn00b: Do you have codec_*.so in /usr/lib/asterisk/modules? Do you have a modules.conf file in /etc/asterisk? |
| 14:57.17 | ManxPower-work | shamelessn00b: I doubt that message is the actual error. I suspect the actual error is elsewhere. |
| 14:57.34 | jsmith | hippiehacker: I doubt the card is fried, as the card is being seen and it's out of alarm. |
| 14:58.02 | ManxPower-work | jsmith: but without seeing the zapata.conf files we don't know much. |
| 14:58.23 | jsmith | hippiehacker: Can you also pastebin /etc/zaptel.conf? Everything looks OK in your zapata.conf file, at least at first glance |
| 14:59.02 | shamelessn00b | http://pastebin.com/m5ea8385c |
| 14:59.17 | shamelessn00b | thats the messages I get on my asterisk when the call is established |
| 14:59.25 | hippiehacker | jsmith: done |
| 14:59.39 | jsmith | shamelessn00b: Again, that's because Asterisk hasn't loaded the codec modules. Did you check the two things I asked you to previously? |
| 14:59.56 | shamelessn00b | jsmith: let me check |
| 14:59.56 | jsmith | shamelessn00b: You can also check the output of "module show" for any codec modules |
| 15:00.20 | ManxPower-work | [Dec 31 19:58:05] WARNING[9946]: file.c:991 ast_streamfile: Unable to open /root/main_menu (format 0x4 (ulaw)): No such file or directory |
| 15:00.51 | shamelessn00b | usr/lib/asterisk/modules/codec_alaw.so /usr/lib/asterisk/modules/codec_dahdi.so |
| 15:00.52 | jsmith | hippiehacker: OK, just to make sure... you've got a partial PRI with the D channel on channel 12? |
| 15:00.56 | Katty | hi manx |
| 15:01.06 | shamelessn00b | jsmith: the files exist |
| 15:01.06 | jsmith | shamelessn00b: That's it? No more than that? |
| 15:01.10 | shamelessn00b | yeah |
| 15:01.17 | jsmith | shamelessn00b: You should have a *lot* more |
| 15:01.19 | ManxPower-work | Hello Squirrel Lady! |
| 15:01.35 | shamelessn00b | so should I recomplie my asterisk?? |
| 15:02.15 | jsmith | shamelessn00b: During the "make menuselect" step, make sure you have all the codecs selected. At a minimum, make sure you get ulaw, alaw, and gsm. |
| 15:02.25 | jsmith | shamelessn00b: Also make sure you get the file format drivers as well... |
| 15:02.46 | shamelessn00b | ok |
| 15:03.01 | ManxPower-work | are you running Asterisk as "root", shameless? |
| 15:03.12 | shamelessn00b | yes |
| 15:03.29 | ManxPower-work | do a ps -aux to confirm |
| 15:03.56 | shamelessn00b | yeah |
| 15:04.01 | shamelessn00b | root owns the process |
| 15:04.07 | jsmith | hippiehacker: Have you install libpri on this system? It's almost like Asterisk isn't seeing the libpri library |
| 15:04.14 | hippiehacker | jsmith: yes, it's worked before, verified with the co |
| 15:04.15 | ManxPower-work | how did you create that main_menu file? |
| 15:04.36 | shamelessn00b | the file plays if I initiate a dahdi call |
| 15:04.38 | shamelessn00b | to my cell phone |
| 15:04.39 | hippiehacker | jsmith: i'll poke around for libpri |
| 15:04.40 | ManxPower-work | hippiehacker: if a kernel update happened or if a file got corrupted..... |
| 15:04.45 | shamelessn00b | so I doubt its the codecs :S |
| 15:04.50 | ManxPower-work | shamelessn00b: That was not my question |
| 15:05.11 | shamelessn00b | main_menu file is 8K 16 bit wav |
| 15:05.20 | shamelessn00b | recorded then converted |
| 15:05.23 | ManxPower-work | mono or stereo? |
| 15:05.24 | dlu_ | all a happy new year |
| 15:05.25 | shamelessn00b | mono |
| 15:05.27 | jsmith | shamelessn00b: It's definitely a problem with codec modules not being loaded |
| 15:05.38 | jsmith | shamelessn00b: The output from "core show translation" is a dead give-away |
| 15:05.50 | shamelessn00b | I can hear it playing over dahdi calls whats your point ManxPower-work |
| 15:05.53 | shamelessn00b | yeah |
| 15:06.00 | shamelessn00b | Im reinstalling asterisk now |
| 15:06.49 | ManxPower-work | shamelessn00b: My point is that you have one of the most bizarre problems I've seen in a long time so don't assume ANYTHING. |
| 15:06.57 | voipmonk | :) |
| 15:07.02 | shamelessn00b | :P |
| 15:07.13 | shamelessn00b | I always face bizarre problems |
| 15:07.17 | shamelessn00b | ALWAYS |
| 15:08.15 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:08.29 | alrs | shamelessn00b: sorry if I missed the beginning of this. Which distribution are you running? Are some parts of Asterisk installed from source, others from package management? |
| 15:08.49 | shamelessn00b | debian 5 |
| 15:09.12 | Katty | hi tony |
| 15:09.30 | shamelessn00b | actually I didnt compile the asterisk souce myself I just got this box with a problem |
| 15:09.36 | shamelessn00b | so now I have to resolve it |
| 15:09.45 | alrs | shamelessn00b: I've had really good results with Asterisk in Debian just sticking with the packaged versions. |
| 15:10.14 | Katty | there's a conspiracy going on. ryan's truck is still parked in front of the house. he was supposed to be at work 1hr and 20min ago :P |
| 15:10.30 | alrs | shamelessn00b: it's entirely possible that manually compiling Asterisk and dropping it on top of the filesystem is making your situation worse, not better |
| 15:11.34 | ManxPower-work | shamelessn00b: Wow, you're totally screwed. |
| 15:11.40 | ManxPower-work | No wonder you have so many problems. |
| 15:12.38 | shamelessn00b | ManxPower-work: compiling from source works for me |
| 15:12.40 | Katty | infobot: crittercam |
| 15:12.41 | infobot | [crittercam] Katty's broadcast of The Nut House @ http://ustre.am/8H5d |
| 15:12.45 | Katty | ^- squirrlies watching you! |
| 15:12.46 | ManxPower-work | shamelessn00b: apparently not. |
| 15:13.04 | ManxPower-work | Are you sure some idiot didn't install a packaged version of Asterisk? |
| 15:13.08 | shamelessn00b | I didnt install asterisk on this box |
| 15:13.13 | shamelessn00b | that explains a lot |
| 15:13.25 | ManxPower-work | I was referring to the idiot that had the box before you. |
| 15:13.33 | alrs | shamelessn00b: compiling from source works. package management works. letting a build script scatter asterisk parts all over package managed asterisk will cause a meltdown |
| 15:14.03 | alrs | ManxPower-work: Package management isn't idiotic, it's just not for you. |
| 15:14.46 | alrs | ManxPower-work: I don't think that you can say with a straight face that you've got a better handle on Asterisk than Tzafrir does. The end. |
| 15:15.02 | ManxPower-work | alrs: Package management is fine. 1) just don't come to us for support and 2) Packaged Asterisk + compiled from source Asterisk == End of Universe. |
| 15:15.16 | *** join/#asterisk dymaxion (n=dymaxion@78-86-174-224.zone2.bethere.co.uk) |
| 15:15.35 | *** part/#asterisk digitalirony (n=digitali@my.ass.looks.just.like.your-face.info) |
| 15:16.18 | ManxPower-work | alrs: I've been using Asterisk in production since early 2002. I don't say I'm better at it than tzafrir__laptop, but I do have quite a bit of experience with Asterisk. |
| 15:16.35 | *** join/#asterisk digitalirony (n=digitali@my.ass.looks.just.like.your-face.info) |
| 15:17.11 | ManxPower-work | And I have experience here seeing all those poor sods that install Asterisk from a package and then can't find the docs or don't have any kernel modules or channels or whatever else the packager decided to put in different packages. |
| 15:17.51 | tzafrir__laptop | looks for kernel modules in the source tree of Asterisk |
| 15:17.54 | ManxPower-work | And when they uninstall the package and install from source things start working. |
| 15:18.05 | Katty | 'poor sods'? |
| 15:18.30 | drmessano | What about if you install Asterisk from an .msi .. ? |
| 15:18.45 | ManxPower-work | tzafrir__laptop: It it matters, I've never heard of problems with the packages people use with the Astrabank. |
| 15:18.50 | Katty | how about some tea and cookies instead |
| 15:18.51 | alrs | ManxPower-work: I agree, Digium should be working to get Dahdi into mainline |
| 15:18.54 | *** join/#asterisk Akiraa (n=Akiraa@79.112.38.145) |
| 15:19.22 | ManxPower-work | There ARE a few people that could build a decent package of Asterisk, but random people are not those few people. |
| 15:20.33 | ManxPower-work | Other issues with packages are 1) they are usually out of date and 2) for some reason most packagers seem to think the default location of files is not good enough for them. |
| 15:21.08 | drmessano | The best thing Digium has done with Dahdi is remove it's dependency for core asterisk functions |
| 15:22.05 | ManxPower-work | alrs: That I agree with. DAHDI code base is pretty stable and doesn't go thru significant changes all the time. (unlike Asterisk) |
| 15:22.57 | ManxPower-work | Oddly, I actually use packaged Asterisk and Zaptel. 1) I don't ask for support on them here and 2) I built the packages myself with the stuff I need |
| 15:23.41 | Katty | http://www.datacenterknowledge.com/wp-content/uploads/2009/12/wetadigital-470.jpg <- here's the data center behind Avatar |
| 15:24.02 | Katty | ^- 10,000 square foot server farm. |
| 15:24.35 | Chainsaw | Katty: He has a strange purple belt. |
| 15:25.32 | Katty | Chainsaw: that's what happens when you process 17.28 gigs a minute. |
| 15:27.14 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:27.41 | Katty | http://i.imgur.com/FhHpR.jpg <- what, wait? |
| 15:28.01 | Katty | s/wait/what/ s/what/wait/ |
| 15:28.12 | Katty | i'm feeling dyslexic this morning. |
| 15:28.44 | Chainsaw | Katty: Good price for a Blu-Ray. |
| 15:29.51 | Katty | Chainsaw: ref: Only |
| 15:30.27 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-130-71.dsl.stlsmo.sbcglobal.net) |
| 15:30.54 | LemensTS | what would be a good dial pattern for calling usa to canada? 011NXXNXXNXXXX ? |
| 15:31.05 | LemensTS | I only want to allow canada |
| 15:31.18 | ManxPower-work | LemensTS: Calling canada is EXACTLY like calling USA |
| 15:31.31 | ManxPower-work | you'll have to match on a list of Canadian area codes. |
| 15:31.44 | ManxPower-work | All numbers in NANPA are dialed the same. |
| 15:31.59 | LemensTS | You don thave to dial 011 ? |
| 15:32.11 | ManxPower-work | you don't call canada much from the usa do you? |
| 15:32.14 | jsmith | LemensTS: Nope... 19 countries share the country code of "1", including the US and Canada |
| 15:32.23 | LemensTS | Never :) just read on google tho i guess that site was wrong |
| 15:32.28 | ManxPower-work | and all those countries would be dialed as 1+ |
| 15:32.30 | jsmith | LemensTS: It's called the "North American Numbering Plan" |
| 15:32.33 | LemensTS | Cool thanks! |
| 15:32.48 | ManxPower-work | LemensTS: your best option is to have your carrier limit the calls. |
| 15:32.49 | jsmith | localcallingguide.com might help you as well |
| 15:33.04 | ManxPower-work | If you can't do that then you'll have to handle it yourself. |
| 15:33.35 | drmessano | What if someone needs to order beer? |
| 15:33.53 | drmessano | Let's not get crazy here.. Maybe block SOME of Canada |
| 15:35.45 | Katty | neat. |
| 15:35.51 | Katty | i never knew 19 countries shared 1 |
| 15:36.44 | ManxPower-work | Katty: scammers in some of those other 19 countries count on that. |
| 15:37.11 | ManxPower-work | Didn't MEXICO used to be part of NANPA up until the 80s? |
| 15:37.56 | *** join/#asterisk ticoit (n=ticoit@201.191.187.219) |
| 15:38.56 | coppice | Katty I think there are quite a few countries in the Caribbean when you count them carefully |
| 15:39.33 | *** part/#asterisk LemensTS (n=customgt@adsl-70-238-130-71.dsl.stlsmo.sbcglobal.net) |
| 15:39.54 | *** join/#asterisk Akiraa (n=Akiraa@79.112.38.145) |
| 15:40.06 | jsmith | ManxPower-work: I dont' think so. |
| 15:41.19 | coppice | MaxPower-work: mexico is country code 52. You don't get a 2 digit code by asking for one late :-) |
| 15:44.26 | Katty | what's the UK? |
| 15:44.30 | Katty | 44? |
| 15:44.56 | vally | yes |
| 15:44.58 | coppice | Its that bit of land just before an american reaches mainland europe |
| 15:45.24 | Katty | smart tail :P |
| 15:48.30 | Kobaz | so what other countries share the 1 then... as far as i know only mexico, the us, and canada are in north america |
| 15:48.44 | Kobaz | there are those little islands and stuff in the carribean, do they count? |
| 15:50.04 | coppice | most carribean countries have high standards of literacy and numeracy, so they probably count better than the US |
| 15:50.08 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 15:50.09 | jsmith | Kobaz: Yes, they count... think Jamaica, Dominican Republic, etc. |
| 15:50.34 | Katty | Kobaz: http://en.wikipedia.org/wiki/List_of_country_calling_codes#Zone_1_.E2.80.93_North_American_Numbering_Plan_Area |
| 15:51.01 | Kobaz | heh |
| 15:51.21 | Katty | i've never even heard of some of these places before. |
| 15:51.39 | *** join/#asterisk ChannelZ (i=channelz@burner.com) |
| 15:51.47 | ariel_ | hello everyone |
| 15:51.50 | ManxPower-work | "Callers in the North American Numbering Plan were once able to call specific areas in Mexico using area codes 706 (northwest Mexico) or 905 (Mexico City). These were officially removed from service in February 1991 in favour of international dialling using country code 52. " |
| 15:52.03 | jsmith | ManxPower-work: How 'bout that... |
| 15:52.06 | ManxPower-work | http://www.wtng.info/wtng-52-mx.html |
| 15:53.42 | Katty | there's no country code for antartica. |
| 15:53.54 | Katty | or the north pole/artic ocean. |
| 15:54.14 | Katty | or the moon :< |
| 15:54.27 | Katty | just you wait and see, we'll be able to call the moon someday. |
| 15:55.10 | ariel_ | moon, mars should we hope |
| 15:55.11 | jsmith | Yeah, but the latency is killer :-) |
| 15:55.33 | ManxPower-work | Moon $14.50/min. Moon (Cellular) $38.25/min |
| 15:55.58 | Katty | jsmith: they're working on that. |
| 15:56.01 | ariel_ | well we have 9.95 per minute now via sat to ships. |
| 15:56.13 | Katty | jsmith: quantum entanglement theory is helping |
| 15:56.34 | jsmith | can only hope |
| 15:57.28 | Katty | so i was going to run by the grocery store on the way home and get some groceries for this weekend. a thought crossed my mind of getting something sweet to bake...like cookies or cake or brownies or something. |
| 15:57.32 | Katty | suggestions? |
| 15:58.05 | ariel_ | Cookies |
| 15:59.11 | vally | cake |
| 15:59.21 | voipmonk | a gd twinkee |
| 16:00.05 | ariel_ | Actually Nice Chocolate Chip cookies Warm with Ice Cream on top....hummmmm |
| 16:00.08 | SkramX | what's the easiest way to call Skype via SIP? |
| 16:00.15 | jaytee | Twinkies have a shelf life of 50 years |
| 16:00.31 | Katty | ariel_: that does sound good. |
| 16:00.34 | ariel_ | that due to them not being real |
| 16:00.44 | jaytee | SkramX, check out Skype for Asterisk |
| 16:00.54 | Corydon76-dig | jaytee: Actually, they don't |
| 16:01.27 | jaytee | Corydon76-dig, well....maybe not urban legends persist :-) |
| 16:01.36 | SkramX | I know this is #asterisk but I was looking for something asterisk independent |
| 16:01.53 | jaytee | SkramX, Google is your friend |
| 16:02.16 | drmessano | I loved when Jay Leno was busting on Twinkies 75th anniversary "Yep, the Twinkie turned 75 today... so that first batch should be expiring any day now.." |
| 16:02.19 | jaytee | but you won't find much since Skype is a closed protocol |
| 16:02.50 | pigpen | quickie question: If I don't have the app_set= in the asterisk.conf (using 1.6.1.12) does this mean that it is being defaulted to 1.4 ?? (seems like it) |
| 16:02.56 | drmessano | Get in on the SIP for Skype beta |
| 16:03.03 | drmessano | or buy Skype for Asterisk |
| 16:04.12 | drmessano | Anything else will be a some Charlie Foxtrot of a setup running the client on a machine, tethered via some dodgy app that speaks SIP worse than Asterisk 1.0 |
| 16:05.20 | SkramX | yeah |
| 16:07.09 | *** part/#asterisk alrs (n=lars@98.sub-75-214-211.myvzw.com) |
| 16:11.40 | drmessano | I dont even use Skype that much, but for $66, at my going rate of $13 an hour for sitting around scratching my ass and watching Criminal Minds reruns, it gave me 5 more hours of TV that I would have otherwise spent making some of these Skype<>CRAP gateways work |
| 16:11.50 | jblack | rubs his eyes |
| 16:12.09 | jblack | morning cannel |
| 16:12.22 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 16:12.34 | ariel_ | at $ 13.00 per hour your cheap |
| 16:12.50 | drmessano | $13 an hour is the value of your spare time |
| 16:12.55 | jblack | ariel_: Have you heard about the 17% unemployment? |
| 16:13.07 | drmessano | Not what I make an hour or charge an hour |
| 16:14.19 | Katty | someone tipped the tray over! )=< |
| 16:14.26 | drmessano | I read something a few years ago that stated your free time is worth, on average, $13 an hour.. I have used that considerably in how much time I invest in working on something to save a few bucks |
| 16:14.26 | Katty | jblack a cup of hot cocoa. |
| 16:14.34 | drmessano | Its thus far been tremendous for me |
| 16:15.19 | Katty | drmessano: such as? |
| 16:15.46 | pigpen | I would like to know what this "Free Time" thing is.... |
| 16:15.47 | Katty | drmessano: driving an extra 5 miles to save a few dollars on groceries? |
| 16:15.51 | drmessano | You'll blow $30 on mexican food without thinking about it, but spend 4 hours working on some kludge of a home project to save spending the same $30 on a perfectly suitable store bought? |
| 16:16.03 | ManxPower-work | drmessano: Amazing, huh? |
| 16:16.22 | Katty | well. |
| 16:16.27 | Katty | that all depends on whether you enjoy it or not |
| 16:16.36 | ariel_ | well then what is the fun of it if you can't work on it? |
| 16:16.37 | drmessano | I not talking about as a hobby |
| 16:16.39 | Katty | cause you can spend 1 dollar on the value menu |
| 16:16.41 | Katty | or cook something. |
| 16:17.03 | Katty | or in ya'lls case, probably 5 bucks on the dollar menu. |
| 16:17.22 | drmessano | Im talking about calculated cost savings based on the concept of "If I do it myself, it will be cheaper" |
| 16:17.33 | Katty | nods |
| 16:17.37 | Katty | that is very rare. |
| 16:17.44 | drmessano | Actually, yes it is |
| 16:17.55 | drmessano | and if you use the $13 an hour metric, its MIND BOGGLING |
| 16:17.58 | pigpen | drmessano, I agree to a point....you may spend more time in that matter, but you can build something that you can fix if it breaks...and more than likely you will make it better. |
| 16:18.06 | pigpen | But good thought process to weed out the shit stuff. |
| 16:18.16 | ManxPower-work | The thing is Skype for Asterisk IS a closed source thing. |
| 16:18.21 | *** join/#asterisk diatonic (n=diatonic@mail.clearwater-research.com) |
| 16:18.23 | Katty | you are 25% more productive with two screens. |
| 16:18.27 | ManxPower-work | And if it breaks you really can't do much |
| 16:18.36 | drmessano | I've found very little that I could build myself anymore that isnt cheaper to buy |
| 16:18.42 | drmessano | and replace |
| 16:18.45 | Katty | drmessano: i know of one. |
| 16:18.49 | Katty | drmessano: a dog house (= |
| 16:18.52 | drmessano | Children? |
| 16:18.53 | drmessano | Oh |
| 16:18.54 | Katty | drmessano: and not a crappy dog house. |
| 16:18.58 | ariel_ | NO sex.. |
| 16:18.59 | Katty | drmessano: a real, cedar, dog house. |
| 16:19.00 | drmessano | I was gonna say, you can buy them cheap too |
| 16:19.15 | Katty | drmessano: i know for a fact you can build one much cheaper than what you can buy one for. |
| 16:19.28 | pigpen | Depends on the size of the Dog. :) |
| 16:19.35 | Katty | drmessano: because there is a very nice one sitting in the back yard ryan built for riddick, with shingles and everything. |
| 16:19.39 | ManxPower-work | I suspect, if you work the numbers, children are cheaper to buy than to do yourself. |
| 16:19.40 | drmessano | ariel_: Sex is better to buy.. You get exactly what you want, no wasting hours on back rubs, being attentive, etc |
| 16:19.43 | pigpen | I can't buy one big enough. |
| 16:20.04 | Katty | pigpen: we had the same problem, but we eventually did find one. |
| 16:20.07 | Katty | pigpen: it was priced at 3 grand. |
| 16:21.05 | *** join/#asterisk lost_soul (n=noymfb@cpe-74-71-234-100.twcny.res.rr.com) |
| 16:21.38 | pigpen | Katty, Eeeks! I have two large dogs. Even with a nicely built dog house, they still sit in the rain at the back door. Guess I should move it by the door. |
| 16:22.10 | Katty | you leave your dogs outside in the rain? |
| 16:22.14 | Katty | is there something wrong with you?! |
| 16:22.30 | ariel_ | Dogs are made to live outside |
| 16:22.37 | Katty | i'd never leave my dog out in the rain :< |
| 16:22.42 | drmessano | No, they leave themselves out in the rain, apparently |
| 16:22.52 | drmessano | he made a house for them, they choose not to us it |
| 16:22.55 | drmessano | FAIL |
| 16:23.01 | fenrus | my dog likes sitting next to me in the couch while it rains outdoors. |
| 16:23.02 | drmessano | use* |
| 16:23.07 | Katty | if it's drizzling outside and i let him out, he sits and the door and whines until i let him back in |
| 16:23.39 | Katty | but he's been an inside puppy all his life |
| 16:24.03 | drmessano | When it starts raining I throw all my cats outside.. Not sure if they like it or not from the screams and clawing at the door I hear, but it's free water, you know? |
| 16:24.09 | ariel_ | oh it's a whinning dog, my Lab loves outside and if she see's rain she runs in it then goes at her own time into the dog house in the back. But she is free to do what she wants. |
| 16:24.54 | Katty | i have invested a TON of money into my dog. |
| 16:24.59 | Katty | i don't just leave him outside |
| 16:25.43 | jblack | drmessano: they're trying to tell you that they're really happy. |
| 16:25.50 | Katty | http://farm4.static.flickr.com/3474/3916280412_b085d54d8a_b.jpg <- we even go through mcdonalds on the way to the park. |
| 16:26.15 | fenrus | yum |
| 16:26.18 | jblack | You buy your dogs cheeseburgers? That's love |
| 16:26.37 | jblack | YOU ARE SO MEAN! |
| 16:26.48 | Katty | http://farm4.static.flickr.com/3526/3916282012_ccf667912e.jpg <- his favorite napping spot is the love seat. |
| 16:26.52 | Corydon76-dig | It's only love if you share the food with your pet |
| 16:26.52 | jblack | First you strap his jaw shut, then you shove a cheseburger in his face! |
| 16:27.08 | Katty | that's a halti, not a muzzle. |
| 16:27.24 | Corydon76-dig | For instance, I shared a slice of bread with my bird the other night |
| 16:27.26 | Katty | you do realize i'm only like 30lbs more than my dog, right? :P |
| 16:27.44 | Corydon76-dig | I took a bite, she took a bite, I took a bite, she took a bite |
| 16:27.57 | Katty | Corydon76-dig: my mom shares ice cream with her quaker parrot. |
| 16:28.10 | Katty | Corydon76-dig: and tea, and peanut butter, and just about anything the bird seems interested in. |
| 16:28.15 | fenrus | http://www.ifia.se/~fenrus/pictures/inez-web.jpg my precious |
| 16:28.20 | jaytee | Corydon76-dig, what kind of bird do you have? |
| 16:28.31 | Corydon76-dig | blue-fronted Amazon |
| 16:28.34 | Katty | fenrus: awww. |
| 16:28.38 | jaytee | nice! |
| 16:28.48 | Katty | fenrus: i love the little floppy ears and pink nose :> |
| 16:28.51 | *** join/#asterisk nsgn (n=nsgn@71.20.112.123) |
| 16:28.51 | fenrus | =) |
| 16:28.59 | fenrus | she's quite cute (: |
| 16:29.07 | Corydon76-dig | When she was younger, my eldest sister shared a rice cake with our cat |
| 16:29.17 | jaytee | we have a couple double yellow amazons here at the zoo, Pancho and Ozark. very good talkers.... in fact they rarely shut up :-) |
| 16:29.22 | fenrus | i share ice creams with my dog :) |
| 16:29.54 | Katty | fenrus: i can't eat something without the dog wanting a bite. |
| 16:29.56 | jblack | I am so gonna caption that and put it on icashazcheezburger.com |
| 16:30.00 | Corydon76-dig | We're still trying to get Big Momma to talk intelligibly |
| 16:30.21 | fenrus | =) |
| 16:30.27 | Katty | fenrus: if you ever have to give your dog pills, like antibiotics, or antihistamine, or whatver...i highly recommend braunswager. |
| 16:30.39 | jaytee | we used to give my dog ice cream cups whenever the ice cream man came around. my dog would go nuts when she'd hear the ice cream truck's music playing before we could hear it. |
| 16:30.49 | Corydon76-dig | Katty: Velveeta works fine for us, here |
| 16:30.55 | Katty | Corydon76-dig: that's a neat idea. |
| 16:31.05 | Katty | velveta's probably cheaper too |
| 16:31.09 | fenrus | Katty, she eats pills for some hormone difficiency, she has no problems eating it when its mixed with her food =) |
| 16:31.17 | Katty | fenrus: k (= |
| 16:31.29 | Katty | riddick didn't like his antihistamine pills. |
| 16:31.34 | Corydon76-dig | Katty: it's like modeling clay... you can wrap a small amount around any pill, and the dog nearly gulps it down whole |
| 16:31.36 | Katty | but the braunswager worked like a charm. |
| 16:31.46 | Katty | Corydon76-dig: yup, same way with the braunswager. |
| 16:31.50 | fenrus | i usually use meatballs :D |
| 16:32.00 | fenrus | if there's a problem :D |
| 16:32.48 | Corydon76-dig | In fact... I'd use Velveeta for modeling clay before I'd use it for food |
| 16:32.50 | Katty | http://farm4.static.flickr.com/3194/3023014801_1cd858fb31_o.jpg <- old photo of Mom with her two cats... Vern and Ernestine. |
| 16:33.08 | Kobaz | wheres the one with you and two cats |
| 16:33.14 | Katty | i don't have cats. |
| 16:33.35 | ManxPower-work | Katty: You don't have cats? That explains alot. 8-) |
| 16:33.36 | Corydon76-dig | Katty: my parents have Ebenezer and Florence... shortened, Eb and Flo... |
| 16:33.45 | Corydon76-dig | Katty: they live on a tidal estuary |
| 16:34.02 | Katty | http://farm4.static.flickr.com/3074/3023840120_f94eae8f2a_o.jpg <- day i was born. |
| 16:34.10 | Katty | or maybe it was the day after. i forget. |
| 16:34.18 | Katty | she told me once. |
| 16:34.33 | Katty | ^- mid 80s. |
| 16:34.34 | ariel_ | ahh so cute |
| 16:35.38 | Kobaz | whose cute... the baby or the mom? |
| 16:35.39 | Corydon76-dig | I'd have cats, but I'm allergic to pussy |
| 16:35.45 | fenrus | heh, i found the pics i was looking for: http://www.ifia.se/~fenrus/inez/ |
| 16:35.55 | Katty | http://farm4.static.flickr.com/3254/3023842966_6c13da4f07.jpg <- i had a great childhood. |
| 16:36.11 | fenrus | Katty, :) |
| 16:36.38 | Kobaz | where's all the pictures that aren't from the 80s |
| 16:36.48 | Katty | fenrus: you take lovely photos. |
| 16:36.58 | fenrus | Katty, thank you |
| 16:37.05 | Katty | Kobaz: also on flickr |
| 16:37.18 | Katty | Kobaz: what are you looking for, my dog? |
| 16:37.25 | Kobaz | heh |
| 16:37.26 | Kobaz | i dunno |
| 16:37.36 | Katty | are you looking for me all grown up? :P |
| 16:37.42 | fenrus | =) |
| 16:37.46 | Kobaz | sure |
| 16:37.55 | fenrus | i was born in the mid 80's, and im not grown up! |
| 16:38.02 | Kobaz | i already saw one of you hiking in the south somewhere |
| 16:38.22 | ariel_ | are we ever grown up or just at different stages of life? |
| 16:39.14 | jblack | most humans have finished physically maturing in their early 20s |
| 16:39.22 | *** join/#asterisk rare1980_ (n=rare@119.152.132.249) |
| 16:39.59 | nsgn | so any reason why I can call out on my digium analog card but calls in don't even hit the asterisk log in it's most verbose form? |
| 16:40.08 | nsgn | calling out works like a charm |
| 16:40.08 | ariel_ | I hate it when Pandora just stops playing wanting me to press a key.... |
| 16:40.17 | nsgn | set that up no problem. incoming calls just don't freakin hit |
| 16:40.19 | *** join/#asterisk bcrisp (n=bcrisp@70.102.242.138) |
| 16:41.25 | ariel_ | do you have any rules setup for answering the call in your inbound context? |
| 16:41.29 | bcrisp | whats it called when the mic 'booms' when a person says a word starting with a P or B? |
| 16:41.59 | ManxPower-work | bcrisp: that's called "crappy mic" |
| 16:42.08 | bcrisp | ManxPower-work, i think she was just too close to the mic |
| 16:42.13 | ariel_ | is playing the Air Piano to the beat of Coldplay's Clocks........... |
| 16:42.26 | bcrisp | trying to clean it up in audacity |
| 16:43.24 | fenrus | distortion? ;) |
| 16:44.17 | nsgn | ariel_, i should, but even if i didn't it should hit the log that the digium has something ringing in should it not? |
| 16:45.11 | *** join/#asterisk hawkinss (n=hawkinss@pool-98-117-58-61.bltmmd.fios.verizon.net) |
| 16:45.27 | bcrisp | i used dynamic range compression now its golden |
| 16:45.40 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 16:45.40 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:46.36 | Katty | herroes Deeewayne |
| 16:47.24 | eppigy | NEIN |
| 16:48.45 | Katty | ohhaidertoyoutoo |
| 16:49.20 | eppigy | hiya |
| 16:49.26 | nsgn | argh. i dont understand why i can't get a peep of an incoming call |
| 16:49.33 | Deeewayne | Katty, g'day! |
| 16:50.25 | Katty | Deeewayne: and what are you going to do with the very last day of 2009? |
| 16:50.38 | Katty | Deeewayne: i was thinking about going out for long island iced teas. |
| 16:50.41 | ariel_ | argh, my wife is going to kill me. My pen just leaked in the pocket....argh......... |
| 16:50.47 | Katty | uh oh. |
| 16:51.14 | Katty | buy new pants on the way home, throw old ones away :P |
| 16:51.48 | ariel_ | shirt |
| 16:55.29 | drmessano | Wipe some lipstick on it |
| 16:55.34 | drmessano | She'll forget about the pen |
| 16:55.38 | fenrus | now off from work! |
| 16:58.16 | nsgn | checked line with butt set. it does ring from phone company but doesn't hit the console with the most verbosity enabled. digium card. I continue to be able to call OUT just fine on this card. what gives? |
| 16:59.40 | *** join/#asterisk errotan (n=errotan@81.0.115.3) |
| 17:01.01 | *** join/#asterisk etfonhomey (n=etfonhom@74-143-192-74.static.insightbb.com) |
| 17:02.20 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 17:03.28 | Katty | ariel_: oh, well, shirts are even cheaper. |
| 17:04.05 | Deeewayne | Katty, I am considering somersaulting into 2010 |
| 17:04.16 | ariel_ | nsgn: make this easy, make your inbound context, exten => s,1,noop |
| 17:04.26 | ariel_ | you should see something then |
| 17:04.27 | Katty | Deeewayne: oh? |
| 17:04.46 | ariel_ | Katty: yes cheaper and good Idea t/y |
| 17:04.50 | ManxPower-work | You should still see something on the console when a call comes in, Starting Simple Switch on..... |
| 17:05.07 | nsgn | yeah i really should still be seeing something anyway |
| 17:05.51 | nsgn | yet i know the card and lines are functional. i can call out all i want and it works exactly as desired. lines are ringing in, but * just doesnt get the idea that they are. i dont remember doing ANYTHING last time i set one of these digium cards up before i could at least see it in the console |
| 17:05.58 | ariel_ | I wish I could turn off nagios alerts with a reply email.... |
| 17:06.10 | ManxPower-work | nsgn: contact Digium. |
| 17:06.25 | nsgn | any commands i can run at the console to give me a better idea of what it expects on the incoming side of things? perhaps asterisk is somehow not even registering the card for incoming? |
| 17:07.18 | ManxPower-work | nsgn: "zap show channels" |
| 17:07.49 | nsgn | three things listed. pseudo, 1 and 2 |
| 17:07.54 | *** part/#asterisk jsmith (n=nnjsmith@asterisk/training-and-documentation-guru/jsmith) |
| 17:07.58 | etfonhomey | ariel_, what do you have nagios monitoring? |
| 17:08.04 | nsgn | context on 1 and 2 was "from-pstn" |
| 17:08.08 | ManxPower-work | nsgn: make sure your /etc/asterisk/logger.conf is setup to log what you want to the console. |
| 17:08.20 | ManxPower-work | nsgn: and you have an s,1,Whatever in [from-pstn] |
| 17:08.24 | ariel_ | etfonhomey: all our Gateways on our ships. |
| 17:08.26 | Katty | http://www.whitecastle.com/promotions/hangover/coupon <- free coffee at white castle tomorrow. |
| 17:08.31 | nsgn | i have 8 analog pstn lines. should i be showing all 8 there? |
| 17:08.33 | nsgn | seems i should.. |
| 17:08.38 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 17:08.50 | ManxPower-work | nsgn: yes, it should. You don't have them all configured. |
| 17:08.57 | ariel_ | have not seen a white castle in years |
| 17:09.15 | ManxPower-work | If ztcfg -vvv shows all the channels, then the config problem is i /etc/asterisk/zapata.conf |
| 17:09.35 | nsgn | checking into that. thanks! just a sec |
| 17:09.45 | ariel_ | or if dahdi in your chan_dahdi.conf |
| 17:10.32 | nsgn | yeah, was gonna say ztcfg is not found |
| 17:11.50 | nsgn | ah, for some reason it was set to just be a 2 channel card |
| 17:11.54 | nsgn | fixing, reloading, trying |
| 17:13.09 | nsgn | though zap show channels only shows the two still. do i need to rebuild something that bases on the chan_dahdi.conf? |
| 17:13.32 | ariel_ | bbl lunch |
| 17:14.55 | Katty | i should start packing up...we're closing in about 30 min |
| 17:17.25 | bcrisp | lucky u |
| 17:21.04 | Katty | yep |
| 17:21.21 | etfonhomey | I thought we were closed but then saw my boss on IM. |
| 17:22.22 | *** join/#asterisk smooth_penguin (n=smoove@59.95.51.206) |
| 17:26.31 | *** join/#asterisk dymaxion (n=dymaxion@78-86-174-224.zone2.bethere.co.uk) |
| 17:27.50 | Katty | fenrus: i like 9713 the most (= |
| 17:28.15 | Katty | fenrus: the expression in 2291 is the cutest |
| 17:36.15 | *** join/#asterisk catojo (n=catojo@189.24.4.110) |
| 17:36.59 | *** join/#asterisk sp3 (n=sp3@75-144-110-147-Indianapolis.hfc.comcastbusiness.net) |
| 17:37.28 | Katty | the squirrels have been very busy this morning. the feeders are empty. |
| 17:41.00 | rbd_ | I have a situation where a SIP call comes into my asterisk box, goes out to another asterisk box, then comes back to mine. my asterisk box detects a SIP spiral (rightfully so), and uses a Local channel (something like "Now forwarding SIP/10.14.25.240-00000000 to 'Local/1099@public' (thanks to SIP/10.10.140.30-00000001)") ... problem is, this message coming back has important data in the SIP header that I would otherwise want to extract |
| 17:41.01 | rbd_ | <PROTECTED> |
| 17:41.49 | Katty | http://i.imgur.com/qecNY.jpg <- awesomeness. |
| 17:47.44 | SuPrSluG | priceless |
| 17:48.49 | ManxPower-work | rbd_: prefixing variables with a single underscore (_) or a double underscore will make child channels inherit the variable. I don't think it would be transported in your case ragardless, but it's something to try |
| 17:49.07 | *** join/#asterisk dymaxion (n=dymaxion@78-86-174-224.zone2.bethere.co.uk) |
| 17:49.58 | rbd_ | MaxPower: I'll try it out, but I've noticed the Local channel doesn't have a /n suffix on it (so afaik it won't migrate the channel variables over...?) |
| 17:50.24 | *** part/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
| 17:50.41 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 17:51.34 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 17:53.40 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.143.43.dsl.dyn.forthnet.gr) |
| 17:57.35 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 18:00.12 | Katty | well in case i don't get to talk to you guys before tomorrow |
| 18:00.20 | Katty | have a lovely evening, and try not to over-do it |
| 18:00.38 | VxJasonxV | How often should Asterisk be re-resolving a SRV endpoint for SIP UDP registration. |
| 18:00.42 | VxJasonxV | ? |
| 18:00.47 | Kobaz | overdoes it |
| 18:00.59 | Katty | detaches, goes home. |
| 18:05.22 | *** join/#asterisk shamelessn00b (n=itsusama@119.160.108.243) |
| 18:05.34 | shamelessn00b | http://pastebin.com/m539aa2ae |
| 18:05.49 | shamelessn00b | copy the text in your notepad |
| 18:05.58 | shamelessn00b | then replace all 6s with underscores |
| 18:11.36 | VxJasonxV | cute |
| 18:12.30 | bcrisp | hey shamelessn00b saw you in csharp heh |
| 18:12.45 | shamelessn00b | :) |
| 18:13.00 | shamelessn00b | yeah, I used to be active there earlier half of this year |
| 18:13.13 | shamelessn00b | was busy making my own mini asterisk |
| 18:13.14 | shamelessn00b | :) |
| 18:13.18 | shamelessn00b | in c# |
| 18:13.57 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek) |
| 18:17.00 | *** join/#asterisk rare1980_ (n=rare@203.175.76.220) |
| 18:18.25 | *** join/#asterisk catojo (n=catojo@189.24.4.110) |
| 18:19.00 | *** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk) |
| 18:19.43 | VxJasonxV | I was having a problem last night where I was getting a username digest mismatch error, at the time, it was happening on register. Now, it's happening when users call me via the external SIP provider. |
| 18:20.15 | *** join/#asterisk sahafeez (n=sahafeez@65-119-47-100.dia.static.qwest.net) |
| 18:20.46 | VxJasonxV | I kinda wonder if my soft phone is just being weird... |
| 18:22.18 | VxJasonxV | looks like it's just my softphone. if I open the official Gizmo client and call myself, it... well, it still errors (extension not found), but it doesn't have this authentication error. |
| 18:23.33 | *** join/#asterisk etnos (n=etnos@adsl-144-252-232.mia.bellsouth.net) |
| 18:25.47 | shamelessn00b | VxJasonxV do you have the extension your dialing defined in the dialplan |
| 18:25.48 | shamelessn00b | ? |
| 18:25.57 | shamelessn00b | extensions.conf |
| 18:26.07 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:26.31 | nsgn | in chan_dahdi.conf how would I go about making two groups of the 8 channels available on my digium card? i have certain lines i want able to call out from a set of phones, and others restricted |
| 18:28.05 | shamelessn00b | when you run the script wancfg_dahdi |
| 18:28.09 | shamelessn00b | it asks you which group |
| 18:28.23 | shamelessn00b | you want to have for that pri right? |
| 18:28.57 | shamelessn00b | can you pastebin the dahdi_chan file |
| 18:28.57 | shamelessn00b | ? |
| 18:29.17 | VxJasonxV | well, that's the thing, for right now, I just want external calls to go to a menu |
| 18:29.27 | nsgn | it's really simple. has context from-zaptel, group=0, channel=1-8 |
| 18:29.29 | VxJasonxV | so, I have a s, and a _., definition with some cute playback executions |
| 18:30.54 | shamelessn00b | exten =>1234,n,answer() |
| 18:31.11 | [TK]D-Fender | nsgn: group=0, channel=1-4 , group=1, channel=5-8 , |
| 18:31.20 | shamelessn00b | exten => 1234,n,goto(some context) |
| 18:31.21 | VxJasonxV | blah, I may have contexts messed up |
| 18:31.40 | shamelessn00b | define that in your default context |
| 18:31.49 | VxJasonxV | and, I did. |
| 18:31.58 | *** join/#asterisk p3nguin (i=gpz5GvdF@9xf6gbZ.a2infotech.com) |
| 18:32.26 | shamelessn00b | you can also try and define a context like 1234 |
| 18:32.37 | shamelessn00b | and include it in your default context |
| 18:32.53 | shamelessn00b | exten => include(1234) |
| 18:33.05 | shamelessn00b | and you will have a context [1234] |
| 18:33.08 | shamelessn00b | down below |
| 18:33.32 | *** join/#asterisk nny (n=scott@64.203.239.83) |
| 18:33.37 | [TK]D-Fender | [13:32]<shamelessn00b>exten => include(1234) <-- Thats.... creative... |
| 18:35.03 | Tim_Toady | lul |
| 18:35.03 | nny | hmm about to test this on my SPA962, but working on a quote for 15 Cisco Phones (509g). The client wants to share the line appearance between all phones. I am wondering about the limitations of this, or if it would prevent each phone from having a unique extension that could be transfered to. Any thoughts appreciated |
| 18:35.03 | *** join/#asterisk hRedBeard (n=hkb@vtelinet-209-134-40-212.vermontel.net) |
| 18:35.03 | shamelessn00b | lol |
| 18:35.03 | hRedBeard | hey everyone |
| 18:35.03 | shamelessn00b | I meant |
| 18:35.04 | shamelessn00b | include => 1234 |
| 18:35.04 | VxJasonxV | I'm actually kinda confiused of how this works. I'm using Gizmo for SIP termination, I want to use it both ways. |
| 18:35.05 | shamelessn00b | sorry about that |
| 18:35.20 | VxJasonxV | So I have a register => with my gizmo authentication in the [general] context. I'm assumed. |
| 18:35.23 | VxJasonxV | ack |
| 18:35.25 | [TK]D-Fender | nny: the phone supports SLA... * does not. |
| 18:35.27 | VxJasonxV | I assumed that register authentication was for incoming. |
| 18:36.04 | nsgn | [TK]D-Fender, so it just reads them in sequence? |
| 18:36.12 | VxJasonxV | So I created a peer definition for Gizmo, with context=gizmo. So the incoming gizmo calls use the gizmo context, and not the "from-sip" as definited in [general] |
| 18:36.17 | nny | [TK]D-Fender: is this worth a read http://www.asterisk.org/node/48342 ? |
| 18:36.19 | hRedBeard | does anyone have any recommendations for good did providers with numbers in the US and Canada? |
| 18:36.21 | VxJasonxV | Is that intentional? |
| 18:36.37 | hRedBeard | not really finding too many useful resources via google... |
| 18:36.44 | smooth_penguin | hRedBeard, callcentric? |
| 18:36.45 | [TK]D-Fender | nsgn: Set parameters. assign current parameters to channels. Change some values. Assign values to some more channels. |
| 18:39.22 | p3nguin | hredbeard: VoIP.ms |
| 18:39.22 | nny | [TK]D-Fender: unfortunately I would prefer to never have to do SLA, it's a relic from technology that existed as a limitation, not a feature (at least IMHO) |
| 18:39.25 | *** join/#asterisk sahafeez (n=sahafeez@65-119-47-100.dia.static.qwest.net) |
| 18:39.43 | nny | [TK]D-Fender: going to try and converse them out of it, just looking to see what would happen if they decided to do it |
| 18:40.08 | hRedBeard | smooth_penguin, p3nguin: thanks I am looking at them now... |
| 18:40.11 | nsgn | [TK]D-Fender, so tward the end of the file i can just start doing "group=0; channel=1; group=1; channel=2" ? |
| 18:40.59 | nny | it's sad really, the point of it would be "why" someone would want SLA. I'll work on refuting any of the reasons it should exist |
| 18:41.03 | smooth_penguin | :> |
| 18:42.00 | nny | the more I read about SLA the further I want to put myself from it ha |
| 18:43.25 | jaytee | I looked at SLA and then once I got over the nausea I used an alternative approach someone suggested in here. |
| 18:43.25 | nny | ahhaa |
| 18:43.25 | nny | indeed |
| 18:43.50 | nny | well it's a matter of why they want it. If it's to track which inbound number is being dialed, there are more elegant ways of handling that. If it's a matter of transfering calls, thats an obvious one |
| 18:44.12 | nny | jaytee: what was the alternative approach suggested btw? |
| 18:44.15 | jaytee | mine had to deal more with what Nortel called multiple appearance or MARP |
| 18:44.54 | ariel_ | Nortel writes them in there favor not yours |
| 18:45.13 | nny | Nortel is also going bankrupt ha |
| 18:45.20 | jaytee | which was to create a second sip account and program one of the line keys on the phone that I wanted the 2nd appearance on and then have them dial the extension which would dial both sip phones at the same time |
| 18:45.23 | nny | i have some shares on my wall as a reminder |
| 18:45.31 | ariel_ | yes good things sometimes happen |
| 18:45.36 | nny | really cheap wallpaper actually :D |
| 18:45.37 | jaytee | Nortel sold their Enterprise group to Avaya |
| 18:46.04 | nny | jaytee: ahh |
| 18:46.23 | ariel_ | Avaya yet another over priced PBX system |
| 18:46.32 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 18:46.36 | nny | jaytee: alternatively couldn't you do like SPECIALEXTEN => Dial(SIP/100&SIP/200) ? |
| 18:46.52 | jaytee | but they drove themselves out of business with their licensing costs and a business model that made them money in the 80's and 90's but couldn't compete in the '00s. |
| 18:47.17 | nny | my favortie was a local nortel dealer saying he doesn't deal with "computer based phone systems" |
| 18:47.25 | nny | i had to show him what a BCM was :D |
| 18:47.34 | jaytee | nny, your specialexten example is exactly how I do it |
| 18:47.58 | nny | jaytee: i assume the unique account was so that the client knew it was that extension being dialed? |
| 18:48.37 | jaytee | nny, not specifically no |
| 18:49.23 | nny | jaytee: appearance reasons? |
| 18:49.56 | jaytee | like if I have Susan at extension 2005 and I want calls to her phone to also ring on another line key on another phone I'd setup 3005 as the second set and then calls to 2005 would go exten => 2005,1,Dial(SIP/2005&SIP/3005) |
| 18:49.57 | *** join/#asterisk catojo (n=catojo@189.24.4.110) |
| 18:50.14 | nny | ahh |
| 18:50.16 | nny | gotcha |
| 18:50.20 | jaytee | mostly for admin assistants |
| 18:50.52 | ariel_ | I setup macro's do that for me. |
| 18:52.35 | jaytee | ariel_, I use 3 custom macros for dialing, one for standard extensions, one for multiple appearance lines and one for phones that don't have voicmail and need to forward no answer to another line. |
| 18:54.30 | hRedBeard | smooth_penguin: have you beeen happy with CallCentric? Any complaints? |
| 18:54.54 | ariel_ | I use macros for just about all inbound and outbound dialings, including my IVR's |
| 18:55.00 | [TK]D-Fender | nsgn: Corre3ct |
| 18:55.54 | smooth_penguin | hRedBeard, well we use it at work, and they have seem to have pretty good service |
| 18:59.37 | nsgn | [TK]D-Fender, thanks. gonna try it |
| 19:00.10 | nny | so i am thinking agent and queue > SLA, does anyone know of a nice way to manage this ona meta level? Is FOP the best way for this? |
| 19:01.17 | nsgn | [TK]D-Fender, can i list them? ex channel=1,2,6,7,8 |
| 19:01.32 | [TK]D-Fender | nsgn: yes |
| 19:01.50 | [TK]D-Fender | nsgn: one by one if you wish |
| 19:02.05 | nsgn | great. thanks. trying |
| 19:04.00 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 19:04.41 | bcrisp | Having issue with call congestion. Using sip provider (didww) for incoming calls to IVR, caller dials extension which executes Dial() to outgoing SIP provider. http://pastebin.ca/1733311 |
| 19:05.16 | bcrisp | ah crap |
| 19:05.21 | bcrisp | saw a typo in the phone # lol |
| 19:05.48 | *** join/#asterisk TSM (n=the_soft@87-194-32-212.bethere.co.uk) |
| 19:09.26 | bcrisp | ok issue resolved |
| 19:10.59 | *** join/#asterisk Tech_Travis (n=tech_tra@208.179.137.131) |
| 19:18.32 | *** join/#asterisk etnos (n=etnos@adsl-2-204-26.mia.bellsouth.net) |
| 19:20.26 | *** join/#asterisk joako_ (n=joako@opensuse/member/joak0) |
| 19:22.02 | SkramX | I'm interested in people's UX opinions. I'm working with an API that has several fields one can search for an object through (i.e. colors, shape, words, size). Should I ask the user what fields they have, or go through the questions one by one and let them answer "I don't know"? |
| 19:24.35 | bcrisp | hmm |
| 19:24.54 | bcrisp | how bout a filter option |
| 19:25.09 | SkramX | ideally, but that's going to be pretty complicated to code |
| 19:25.14 | bcrisp | why? |
| 19:25.33 | bcrisp | create a search criteria object |
| 19:25.47 | SkramX | i guess i just need to think about the call flow |
| 19:25.53 | bcrisp | thats how i typically do it |
| 19:25.58 | SkramX | can you elaborate? |
| 19:25.59 | *** join/#asterisk verywiseman (n=khaled@unaffiliated/verywiseman) |
| 19:26.07 | bcrisp | for example, i have global search in a CRM |
| 19:26.15 | bcrisp | they can search by customer, opportunity, contacts, notes etc |
| 19:26.23 | SkramX | is this *over the phone*? |
| 19:26.24 | ecrane | how complicated could this ss7 nonsense be..... |
| 19:26.31 | bcrisp | lol |
| 19:26.34 | bcrisp | wrong channel |
| 19:26.45 | bcrisp | sorry... |
| 19:26.48 | SkramX | thought i was talking about webdev? |
| 19:26.58 | bcrisp | nah... thought it was a C channel |
| 19:27.05 | SkramX | oh gross! ;) |
| 19:27.15 | bcrisp | maybe use voice recog |
| 19:27.23 | SkramX | well i am already |
| 19:27.25 | ecrane | bcrisp: You are not in the C channel... |
| 19:27.37 | bcrisp | ecrane, i said "A" C channel |
| 19:38.40 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 19:46.19 | ariel_ | it's that time today to head home. See you all next year and hope everyone has a great and safe New Year. |
| 19:47.37 | ChannelZ | yawns and stretches |
| 19:50.42 | nny | wow |
| 19:50.48 | nny | FOP2 is slightly bad ass |
| 19:54.21 | ChannelZ | hmm there's no flag for VoiceMail to not play a greeting at all? |
| 19:55.51 | *** join/#asterisk nsgn (n=nsgn@71.20.112.162) |
| 19:56.13 | nsgn | so i need mpg123 to run music on hold...but on centos yum install mpg123 says no such package. am i missing something? |
| 19:58.52 | tzafrir__laptop | nsgn, you only need mpg123 if you e.g. want to stream a remote mp3 stream |
| 19:58.58 | [TK]D-Fender | nsgn: And why do you need that? |
| 19:59.01 | tzafrir__laptop | otherwise, just use local files |
| 20:01.57 | nny | hrrrm |
| 20:02.02 | nny | experiencing this exact issue |
| 20:02.03 | nny | http://lists.digium.com/pipermail/asterisk-users/2007-May/188072.html |
| 20:02.22 | nny | queues show up as invalid on a restart, but ok aftr reloadin app_queue.so Anyone know why this is? |
| 20:03.01 | ChannelZ | hmm is there some module dependency? |
| 20:03.21 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 20:03.22 | ChannelZ | like, chan_local needs to be loaded first or something? |
| 20:03.35 | nny | i don't see anything in the logs, i'll ook again |
| 20:03.37 | nny | look* |
| 20:06.07 | [TK]D-Fender | ChannelZ: That makes sense... and looks identical to someone elses problem just earlier this week |
| 20:06.18 | *** join/#asterisk QubeZ (n=qube@64.128.254.34) |
| 20:06.22 | QubeZ | hello all |
| 20:06.41 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 20:06.46 | ChannelZ | hallo |
| 20:06.57 | QubeZ | anyone here have personal experience with multiple asterisk vm's on xen? i am setting up a lab and want to know what resources, particular related to asterisk on xen that I should be reading. Also, timing issues a factor on the vm? |
| 20:07.45 | nny | ChannelZ: winner winner chicken dinner |
| 20:07.49 | ChannelZ | [TK]D-Fender: seems plausible as a reason why reloading the module works - I had a similar problem with voicemail and smdi |
| 20:07.55 | nny | ChannelZ: load => chan_local.so in modules.conf fixed it |
| 20:08.04 | ChannelZ | yay |
| 20:08.27 | [TK]D-Fender | nny: Was it not being loaded, or does the explicit load precede autoload? |
| 20:09.43 | nny | [TK]D-Fender: that is a good question, let me remove it and checj |
| 20:09.44 | nny | check* |
| 20:10.42 | nny | modules show like chan_local.so says loaded |
| 20:10.47 | ChannelZ | my guess is it's not a dependency of the queues module specifically, but the local channels needed by his particular config |
| 20:10.54 | nny | er module |
| 20:10.54 | *** join/#asterisk montu (n=dpearson@mail.databalance.com) |
| 20:11.02 | nny | so it is loaded without the modules.conf entry |
| 20:11.12 | nny | which means it's a timing thing no? |
| 20:11.41 | [TK]D-Fender | nny: that was the direction I intended with my question |
| 20:11.47 | [TK]D-Fender | nny: race condition |
| 20:12.17 | nny | [TK]D-Fender: yes than, it appears so |
| 20:12.17 | nny | I assume this would qualify as a bug |
| 20:12.43 | ChannelZ | do a full reload and see what order they are loading it |
| 20:12.43 | ChannelZ | err in |
| 20:12.55 | [TK]D-Fender | nny: Don't think I'd say the same... but something that might warrant a safer approach to loading... |
| 20:13.08 | nny | gotcha |
| 20:13.59 | *** join/#asterisk Alagar (n=Administ@122.164.33.86) |
| 20:14.23 | p3nguin | When people travel and want to use their VoIP systems on the road (from hotels, motels, etc.), what types of hard phones do people take along? I know not everyone relies on soft phones just because they are going mobile. |
| 20:15.33 | bcrisp | sip mobiles? |
| 20:15.56 | VxJasonxV | usually wifi sip mobile phones |
| 20:16.07 | VxJasonxV | unless they have a usb phone of some manner that plugs into their laptop |
| 20:16.26 | *** join/#asterisk trafalgargirl (n=smcgurna@dsl-173-206-65-32.tor.primus.ca) |
| 20:16.32 | bcrisp | i dont think you need wifii on some nets |
| 20:16.47 | nny | ChannelZ: [TK]D-Fender my testing box lacks a DAHDI channel of any kind, this may be the source of the issue, it looks like my cahn_dahdi.cofn is malformed due to this fact, looking into properly making a dummy file now |
| 20:17.08 | bcrisp | i thought motorolla made a voip mobile |
| 20:17.22 | nny | ahh yes, cahn dahdi, kirk's greatest nemesis |
| 20:19.12 | nny | is it ok to not load chan_dahdi if I don't have the hardware? Sorry noob question heh |
| 20:19.25 | p3nguin | We need to exclude cellular phones which are VoIP capable -- I'm not going to get a new cell phone, but I will get an additional VoIP-only phone. |
| 20:20.13 | trafalgargirl | hi - hoping someone can walk me through having to make an urgent change to a call center |
| 20:20.21 | trafalgargirl | time should have closed it at 3 pm |
| 20:20.28 | ChannelZ | nny, you should still have dahdi_dummy being loaded (unless you're not using MeetMe or anything else that needs it) |
| 20:20.38 | tzafrir__laptop | nny, Dahdi is just a nice game: http://www.educ.uvic.ca/globalarts/pages/artexchanges/countries/indiadeenabandhugame/pages/page_7.html |
| 20:20.48 | p3nguin | trafalgargirl: You'll have to actually state a problem and ask a question before anyone can help you. |
| 20:21.02 | montu | I think the answer is no. The "call on hold" feature for polycom ip 430 phones can asterisk be told to timeout these calls on hold and GoTo()? Appears this is a feature handled completely by the handset. |
| 20:21.03 | jaytee | trafalgargirl, you were here last week with the same issue |
| 20:21.04 | trafalgargirl | sorry - i am new here and still leaning |
| 20:21.08 | nny | tzafrir__laptop: heh |
| 20:21.20 | trafalgargirl | yes and voip monk said it was fixed (he did it himself) |
| 20:21.21 | bcrisp | trafalgargirl, pastebin your dialplan |
| 20:21.28 | bcrisp | oh SNAP! |
| 20:21.28 | ChannelZ | or unplug the phone. CLOSED! |
| 20:21.29 | trafalgargirl | however, i have a call center still open |
| 20:21.46 | nny | ChannelZ: i'll look into loading dahdi_dummy now thanks |
| 20:22.17 | ChannelZ | well we have no idea what it means for your call center to be 'open' or 'closed' and thus can't help you much without some sort of context as to what that means for you |
| 20:22.34 | bcrisp | ChannelZ, she's talking about gotoiftime |
| 20:23.04 | trafalgargirl | we have several queues -- all of which take calls in -- and voipmonks programmed the gotoiftime to open and close for the holiday hours |
| 20:23.18 | ChannelZ | bcrisp: ok but the only answer then is "fix it" without seeing anything |
| 20:23.51 | ChannelZ | pastebin your extensions.conf |
| 20:23.56 | p3nguin | trafalgargirl: Pastebin the entire extensions.conf. |
| 20:23.57 | Corydon76-dig | trafalgargirl: are you sure he didn't only fix it for Christmas Eve and not NYE? |
| 20:24.11 | QubeZ | should I avoid Asterisk on a Linux VM hosted on Xen Server? |
| 20:24.28 | trafalgargirl | http://pastebin.com/m450ec5f5 |
| 20:24.35 | trafalgargirl | we had the dates for both |
| 20:24.37 | Corydon76-dig | trafalgargirl: just change the line specifying Christmas eve to the 31st and reload |
| 20:25.03 | Corydon76-dig | trafalgargirl: it's commented out |
| 20:25.11 | trafalgargirl | voipmonks did that this am |
| 20:25.15 | trafalgargirl | as it did not open |
| 20:25.16 | Corydon76-dig | trafalgargirl: uncomment and reload |
| 20:25.17 | trafalgargirl | i think |
| 20:25.28 | bcrisp | ya.. its commented out |
| 20:25.39 | bcrisp | 7.;exten => 5146693933,n,GotoIfTime(*|*|31|dec?ontario_parks_after_hours_holiday,$ |
| 20:25.41 | p3nguin | If it's commented out, how can you expect it to work? |
| 20:25.46 | nny | ChannelZ: any chance you know of a doc to test chan_dummy? This is a VM, so I suspect deeper issues |
| 20:25.52 | ChannelZ | what are the $ at the end of those lines? |
| 20:26.10 | trafalgargirl | p3nguin - i am sorry if you think i am an idiot - but i am very new and all of my VoIP techs are awol |
| 20:26.15 | Corydon76-dig | ChannelZ: it's the editor chopping off at the end of a line |
| 20:26.25 | p3nguin | pico/nano |
| 20:26.26 | ChannelZ | oh.. pico |
| 20:26.36 | ChannelZ | nny: dahdi show status |
| 20:27.00 | Corydon76-dig | trafalgargirl: when do they send you for DCAP training? |
| 20:27.02 | p3nguin | There's a reason to use -w when loading a file with pico/nano. |
| 20:27.32 | bcrisp | ya if its broken call the people responsible id say |
| 20:27.56 | ChannelZ | "Your current wait time is... 72.. hours.." |
| 20:28.10 | nny | ChannelZ: ok so dahdi dummy is working, I must need some juju in chan_dahdi.conf so the module doesn't hang on loading it. That seems to be the source of my issue |
| 20:28.24 | ChannelZ | wait, it's hanging? |
| 20:28.29 | bcrisp | just change the dialplan so that the callcenter rings the voip guys.. ! |
| 20:28.36 | trafalgargirl | that would be nice |
| 20:28.45 | trafalgargirl | another dumb question |
| 20:28.48 | trafalgargirl | i have never reloaded |
| 20:28.56 | trafalgargirl | i have commented out three of the queues |
| 20:29.00 | nny | ChannelZ: yes, if I reload the modules, it stops at -- Reloading module 'chan_dahdi.so' (DAHDI Telephony) == Parsing '/etc/asterisk/chan_dahdi.conf': |
| 20:29.17 | nny | ChannelZ: what the |
| 20:29.21 | nny | ChannelZ: now it doesn't ha |
| 20:29.35 | p3nguin | waits for the question |
| 20:29.52 | Corydon76-dig | trafalgargirl: btw, the reason it didn't open this morning is that you should swap the two lines that specify hours for the 31st |
| 20:29.54 | ChannelZ | nny I don't think you actually need chan_dahdi.conf since dahdi_dummy has nothing to configure |
| 20:30.25 | nny | ChannelZ: I thought so, but right now something is messing with the load order, if I remove chan_dahdi.conf from the load it works |
| 20:30.30 | trafalgargirl | put the 31 first and then the 1 |
| 20:30.33 | ChannelZ | and dummy is more of a psuedo device just being used for timing, it's not really a useable channel (by you) |
| 20:30.35 | Corydon76-dig | trafalgargirl: or just change the first term of the line you just uncommented to 15:00-08:59 |
| 20:30.40 | nny | ChannelZ: yeah I thought so |
| 20:30.49 | montu | Need some help with the parkedcalls feature. Anyone willing? Asterisk 1.4.21.1 |
| 20:30.57 | p3nguin | montu: Ask the question. |
| 20:30.59 | nny | ChannelZ: but somehow the queues module is being a sad panda when loaded normally |
| 20:31.05 | trafalgargirl | how to i do a reload? |
| 20:31.11 | Corydon76-dig | trafalgargirl: see line 12 versus line 7: http://pastebin.com/m450ec5f5 |
| 20:31.18 | ChannelZ | nny didn't we solve that by loading chan_local ? |
| 20:31.24 | Corydon76-dig | trafalgargirl: asterisk -rx 'reload' |
| 20:31.32 | nny | ChannelZ: yes, but the question arose as to if it was a timing issue |
| 20:31.38 | nny | ChannelZ: or rather, race condition |
| 20:31.40 | montu | Parkedcalls functions but when it ringsback after timeout and the original extension ignores the call I would like it to move on to voicemail for that extension. At present the call is dropped. |
| 20:31.57 | Corydon76-dig | trafalgargirl: or more precisely: asterisk -rx 'dialplan reload' |
| 20:32.12 | trafalgargirl | at my command line? |
| 20:32.17 | ChannelZ | nny sure as a curiosity |
| 20:32.17 | Corydon76-dig | Correct |
| 20:32.26 | ChannelZ | nny but if you have it working... :) |
| 20:32.32 | nny | ChannelZ: indeed |
| 20:33.07 | nny | ChannelZ: i'll move on for now, i wanted to see which module was loading first, but I run out of space in my terminal session from console. I'll have to have asterisk log it all |
| 20:33.21 | ChannelZ | do you use screen? |
| 20:33.27 | nny | ChannelZ: no but I should :) |
| 20:33.44 | trafalgargirl | i am getting command not found |
| 20:33.45 | ChannelZ | yes yes |
| 20:34.00 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 20:34.13 | nny | I am working on multiple things at once, both figuring out the module load order issue and forcing myself to use screen regularly are now on my todo list :D |
| 20:34.27 | nny | i have only used screen for training someone |
| 20:34.50 | ChannelZ | montu: does your dialplan actually have people directing to voicemail when someone doesn't answer normally? |
| 20:34.52 | nny | and my feeble gray matter has forgotten all the commands, anyways, it's on the list :) |
| 20:35.14 | ChannelZ | I <hump> screen |
| 20:35.18 | trafalgargirl | i am at my root prompt |
| 20:35.22 | trafalgargirl | should i be somewhere else? |
| 20:35.59 | p3nguin | montu: If it's calling an extension, it will do whatever you tell it to do, since that is what extensions do. |
| 20:36.26 | montu | ChannelZ: for incoming calls yes this uses a macro_dial-ext |
| 20:36.49 | p3nguin | montu: If you meant it is calling a device rather than an extension, then you'll have to check to see what is actually happening by increasing verbosity during the operation. |
| 20:37.22 | ChannelZ | montu: but for each person's EXTENSION - whatever it is that you are telling Park() to dial back to |
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| 20:55.39 | ChannelZ | hmm this is wierd. anyone Dial() a mixture of SIP and IAX? (as in Dial(SIP/foo&IAX2/fee) ?) |
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| 20:56.36 | ChannelZ | meh |
| 20:56.51 | ChannelZ | Take 2 post-netslipt: anyone Dial() a mixture of SIP and IAX? (as in Dial(SIP/foo&IAX2/fee) ?) |
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| 21:00.29 | ChannelZ | ah nevermind different problem |
| 21:00.44 | KaneHau | aloha... ok, I FINALLY got the system to place an outgoing call. However, after it dials the number it does not wait for the phone to be picked up, but instead simply goes on with the rest of the script. Is there something I need to put in to make it wait for the phone to actually be picked up? |
| 21:01.20 | KaneHau | I'm using a call file to initiate the call, and then passing control to a [outgoing] exten |
| 21:03.00 | ChannelZ | well it doesn't necessarily know when the remote end picked up |
| 21:03.10 | ChannelZ | (if by which you mean 'answered the phone') |
| 21:03.26 | *** join/#asterisk bcrisp (n=bcrisp@70.102.242.138) |
| 21:03.27 | KaneHau | yes, my phone rings once and the script blunders ahead with "hello-world" |
| 21:03.34 | KaneHau | before I can even pick it up |
| 21:03.58 | bcrisp | muahaha |
| 21:04.04 | KaneHau | how do I make it wait for the phone to be picked up? |
| 21:04.25 | bcrisp | pastebin your dialplan |
| 21:04.46 | bcrisp | (unless you already have... had to reboot since my machine went nuts) |
| 21:05.06 | trafalgargirl | bcrisp did you send me a pastebin earlier? I lost my connection |
| 21:05.13 | KaneHau | actually, it's only 3 lines: |
| 21:05.14 | KaneHau | [outgoing] |
| 21:05.15 | KaneHau | exten => 1,1,Playback(hello-world)exten => 1,n,Hangup |
| 21:05.22 | bcrisp | send you a pastebin? no |
| 21:05.44 | bcrisp | KaneHau, try exten => 1,1,Answer(), then 1,n,Playback.... |
| 21:05.55 | bcrisp | add delay if needed |
| 21:05.57 | KaneHau | well, that last line is actually 2 lines |
| 21:05.58 | KaneHau | bcrisp: trying, thanks |
| 21:07.14 | trafalgargirl | oh someone did - about the order of my lines |
| 21:07.18 | trafalgargirl | i think |
| 21:07.21 | ChannelZ | you will have to devise a way to detect when someone has picked up |
| 21:07.56 | KaneHau | bcrisp: no change, still blunders on without waiting for the phone to be picked up |
| 21:08.10 | bcrisp | add delay |
| 21:08.19 | bcrisp | core show application Playback |
| 21:08.55 | trafalgargirl | do you think i need to make changes to the order to make sure the queues do not open i the am? |
| 21:09.16 | bcrisp | hmm |
| 21:09.19 | ChannelZ | AMD might be able to help you, or you have to WaitForSilence() for more time than between rings, and hope the person shuts up for that long after they pick up... |
| 21:09.39 | bcrisp | KaneHau, Answer(delay) |
| 21:09.40 | p3nguin | When you originate a call from asterisk, one phone is called first, it has to answer, then the second phone is called. |
| 21:10.58 | p3nguin | originate SIP/18004444444@myitsp extension 1@mycontext |
| 21:11.16 | KaneHau | p3: huh? I'm using asterisk to only dial an outside number and deliver a spoken message, there are not 2 phones (this is DAHDI/g0/phonenumber) |
| 21:11.21 | p3nguin | 18004444444 needs to answer before 1@mycontext executes. |
| 21:11.51 | ChannelZ | He's making a robocaller |
| 21:12.36 | bcrisp | ah |
| 21:13.01 | ChannelZ | He needs to know when the person he dialed actually picked up the phone, but depending on the channel you're dialing that's not possible to know without actually listening and trying to figure it out |
| 21:13.07 | p3nguin | oh, sorry... 1@outgoing |
| 21:13.21 | p3nguin | I missed where you specified a context. |
| 21:13.47 | ChannelZ | IE if I dial a DAHDI channel on a POTS line, * think it 'answered' as soon as it's done dialing the number. |
| 21:14.04 | p3nguin | The defined problem was that the playback started immediately, right? |
| 21:14.09 | KaneHau | egads |
| 21:14.21 | ChannelZ | yes.. before the person picks up the phone. |
| 21:14.26 | KaneHau | p3: the problem is as ChannelZ said... it rings once and then goes immediately on with the rest of the script |
| 21:14.30 | KaneHau | before the phone is picked up |
| 21:14.33 | ChannelZ | It's like me calling you and starting to talk while it's still ringing. |
| 21:14.42 | KaneHau | yup |
| 21:15.57 | p3nguin | Is AMD compatible with DAHDI channels? |
| 21:16.25 | KaneHau | Answer(50000000) isn't delaying at all |
| 21:16.36 | ChannelZ | it's basically an audio analyizer, it should be compatible on any channel so long as * is in the media stream |
| 21:16.38 | KaneHau | What is AMD (other than a competitor to Intel) |
| 21:16.50 | ChannelZ | Answering Machine Detection |
| 21:16.54 | KaneHau | ah |
| 21:16.55 | ChannelZ | see amd.conf |
| 21:17.23 | KaneHau | I would have thought that the Digium boards would have had pickup-detection in the hardware |
| 21:18.35 | KaneHau | I dropped in 10 "hello-world" and it delayed enough for me to pick up the ring and hear it |
| 21:18.38 | KaneHau | gaaaaaa |
| 21:20.55 | nny | anyone use/used asteriskcc (crm) |
| 21:24.52 | KaneHau | I tried AMD with the default settings... and it thought a 'ring' was a spoken word (apparently) |
| 21:25.15 | KaneHau | AMD: Word detected. iWordsCount:1 |
| 21:26.24 | bcrisp | "ring ring ring" :) |
| 21:27.10 | KaneHau | more like "ring" "hello world... hello world.... hello world... hello world" |
| 21:33.25 | KaneHau | well, I put a Wait(15) in, and that does wait 15 seconds before doing the Playback |
| 21:33.46 | KaneHau | but I think that will be confusing for the quick answerer |
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| 21:37.23 | KaneHau | is there perhaps some channel status I can monitor that will tell me if it is picked up or not? |
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| 21:38.44 | nny | woo |
| 21:39.10 | nny | so switchvox uses astercc (crm) it seems. I hope they have updated the software from the version publicly available |
| 21:39.22 | KaneHau | :sha |
| 21:39.25 | KaneHau | whoops |
| 21:40.19 | KaneHau | hmmm WaitForSilence() is interesting |
| 21:41.36 | nny | hmm maybe they don |
| 21:41.37 | nny | 't |
| 21:41.47 | nny | hte one in the switchvox product video looks a lot cleaner |
| 21:42.41 | nny | ahh yeah it's switchbox's switchboard |
| 21:42.53 | nny | kudos to them for making it, heh |
| 21:43.37 | nny | and as a note astercc is scary |
| 21:44.31 | nny | has potential though |
| 21:48.16 | nny | be cool if it wasn't only available for switchvox, but that's the price to pay heh |
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| 21:51.31 | KaneHau | in * CLI, is there a command to print out a variable? e.g., I want to see what DIALSTATUS is during a call |
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| 22:09.30 | nsgn | ok. this is frustrating. i have an extension with no phone registered on it. the extension has been set up with "call forward unconditional" to a cell phone. just dialing the extension does what it should and forwards to the cell...but an IVR forwarding to the extension gets a busy signal |
| 22:09.53 | nsgn | why does a call coming to an extension from an IVR treat the extension any different than just calling it from a sip handset? |
| 22:10.14 | nsgn | it seems to care that there's not a phone registered actively on it, ignoring the call forward configured for the extension |
| 22:10.18 | voipmonk | nsgn: the first step is to look at the debug and see what asterisk tells you. |
| 22:11.04 | nsgn | doing so right now with cli set to verbosity 4. stand by |
| 22:11.18 | voipmonk | verbosity, eh? |
| 22:11.47 | voipmonk | separate these two ideas |
| 22:12.02 | voipmonk | you have verbosity and then you have debug |
| 22:12.17 | voipmonk | ever wonder why you can set the verbosity and then still set debug? :) |
| 22:12.21 | voipmonk | there's a reason for that. |
| 22:12.52 | nsgn | this did what i needed |
| 22:13.08 | nsgn | i check verbosity first cause it spits what it's doing literally back out at me |
| 22:13.19 | voipmonk | thats only 1/2 the picture |
| 22:13.23 | nsgn | in this case i just discovered an error in my ivr that ended up being the issue. i'm a moron :) |
| 22:13.36 | nsgn | but thanks for getting me thinking, however stupid it was |
| 22:13.36 | voipmonk | you have to enable debug as well and set /etc/asterisk/logger.conf as well |
| 22:14.17 | nsgn | yeah. it's been a long day of writing complex crap for this phone system. should have dug that one deeper before bothering here |
| 22:14.21 | nsgn | thanks for the time though |
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| 23:20.15 | nny | wow |
| 23:20.18 | nny | loving fop2 |
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| 23:50.24 | joako | fop2 is new? |
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