| 00:19.05 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:19.05 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:44.01 | ghost75 | what is the AMI command to get voicemails? |
| 00:45.15 | WIMPy | manager show commands |
| 00:45.29 | ghost75 | yes i cant find any |
| 00:46.19 | WIMPy | manager show command mailbox<tab> |
| 00:47.00 | ghost75 | yes i know them |
| 00:47.43 | ghost75 | this is only returning number of messages |
| 00:48.54 | WIMPy | And what do you want? |
| 00:49.03 | ChannelZ | hot wings |
| 00:49.16 | ghost75 | chicken wings |
| 00:49.29 | WIMPy | setz light to ChannelZ. |
| 00:49.34 | WIMPy | There you go. |
| 00:49.42 | ghost75 | i want to see message itself and from whom it is |
| 00:50.00 | WIMPy | You have to read the .txt files. |
| 00:50.46 | ghost75 | where are those store |
| 00:50.48 | WIMPy | Far more interesting would be the question how to delete a message. |
| 00:50.49 | ghost75 | +d |
| 00:51.22 | WIMPy | spool directory |
| 00:53.19 | *** join/#asterisk ghostmediapro (~Trupsalms@unaffiliated/ghostmediapro) |
| 00:53.56 | ghostmediapro | i've setup asterisk to user and device mode, so the users here in the family can log in and out of phone, i want to enable chanspy for the user instead of the device for monitoring the kids |
| 00:54.12 | p3nguin | Why can't you delete the sound and text files? |
| 00:54.35 | WIMPy | I can. But I don't know what Asterisk makes of it. |
| 00:54.44 | ghost75 | asterisk ami can delete file? |
| 00:54.46 | p3nguin | ghostmediapro: We've never heard of "user and device mode." |
| 00:54.52 | p3nguin | I just delete the files. |
| 00:54.53 | WIMPy | Do I have to rename all files? What about race conditions? |
| 00:55.35 | p3nguin | Oh, I don't know what will happen if you would delete, say, message 0 only. That might cause some problems. |
| 00:55.46 | WIMPy | I wanted to do a visual voicemail thing for a long time, but it seems a little dodgy. |
| 00:56.02 | ghost75 | i know a webtool which can do it |
| 00:56.11 | ghost75 | but its all encrypted php :< |
| 00:56.22 | WIMPy | One that can do it or one that can do it most of the time? |
| 00:56.48 | ghost75 | i didnt tested to delete, just saw the option |
| 00:57.20 | ghost75 | http://voip-manager.net/asterisk-phonebook.php |
| 00:57.22 | WIMPy | Might get interesting if you do it while a new message is being recorded. |
| 00:58.20 | ghostmediapro | p3nguin: where you can configure a sip device to register with asterisk, but users 1-5 can sign in and out of that device |
| 00:58.48 | p3nguin | Never heard of it. |
| 00:58.57 | p3nguin | Maybe you're talking about something else. |
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| 01:00.03 | ghost75 | open the txt file should be possible with getconfig but how to delete |
| 01:00.25 | WIMPy | You definitely have to do it yourself. |
| 01:00.36 | p3nguin | You can "open" the text file with any text editor. You can delete the file with rm. |
| 01:00.56 | ghost75 | ami has no rm :) |
| 01:01.28 | WIMPy | The language you use to connect to AMI probably has. |
| 01:02.04 | ghost75 | there must be a way using ami |
| 01:02.24 | WIMPy | I whish there was an official way. |
| 01:05.02 | ghost75 | question like here http://lists.digium.com/pipermail/asterisk-users/2007-March/183309.html |
| 01:07.26 | ghost75 | that tool from above ... i just deleted one voicemail |
| 01:07.54 | ghost75 | you can even listen them |
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| 01:21.50 | ghost75 | how to run system cmd in cli? |
| 01:22.52 | ghost75 | ah ! it was |
| 01:23.08 | ghost75 | this is how they did it: http://phpagi.sourceforge.net/phpagi22/api-docs/__examplesource/exsource_ome_phpagi_devel_phpagi_examples_sip_show_peer.php_a884030dbf98b0261079f0d0ff35ab7b.html |
| 01:26.10 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@190.167.114.81) |
| 01:27.10 | WIMPy | I don't see that doing anything. Just getting data and not using it? |
| 01:27.44 | ghost75 | only example |
| 01:28.37 | ghost75 | with ami you just run system cmd (rm whatever) |
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| 01:59.32 | greenwolf | hi |
| 02:05.08 | SeRi | hi |
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| 03:19.36 | dijib | SeRi: around? |
| 03:21.52 | dijib | anybody around? |
| 03:25.38 | *** join/#asterisk josefig (~josef@unaffiliated/josefig) |
| 03:49.08 | dfgas-cr48 | nope :P |
| 03:56.17 | dfgas-cr48 | dijib, um, what happened to my dialplan? |
| 03:56.39 | dfgas-cr48 | do you have a copy of what you sent me in pm lastweek? |
| 03:57.20 | dfgas-cr48 | i don't as that computer was redone over the weekend (desktop that that was pm'ed to) |
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| 04:48.28 | dijib | dfgas-cr48: yo |
| 04:48.36 | dfgas-cr48 | yo |
| 04:48.36 | dijib | what do you mean what happend to it? |
| 04:48.42 | dijib | did i kill it? |
| 04:48.47 | dfgas-cr48 | i think so |
| 04:48.56 | dfgas-cr48 | so for now i put the old one there |
| 04:49.09 | dijib | eh actually ive had to revert back to 1.8.18... that 11.1.0 was too buggy! |
| 04:49.12 | dijib | did it last night |
| 04:49.24 | dijib | mine started to crash on conf. |
| 04:49.43 | dijib | and would lose registration without any timout errors |
| 04:49.50 | dfgas-cr48 | the one you pm'ed me i got working right but i think you went in there at the same time and save it right after me |
| 04:50.06 | dfgas-cr48 | i am not having those issues at all :( |
| 04:50.23 | dfgas-cr48 | are you able to call up log on pm you sent me by chance or no? |
| 04:50.27 | dijib | so re-write it, you sould have enough to work on. |
| 04:50.41 | dijib | call up who? |
| 04:51.03 | dfgas-cr48 | right now when the wrong number is entered it play the nember is not in service and hangs up |
| 04:51.22 | dfgas-cr48 | do you log your pm's? |
| 04:57.16 | *** join/#asterisk nanoha-sama (~nanoha-sa@2001:470:e97f:1003:215:5dff:fe07:4806) |
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| 05:03.09 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 05:03.09 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.0.1 (2012/11/05), 10.10.0 (2012/11/06), 1.8.18.0 (2012/11/06), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.13 (2012/10/09) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 05:04.57 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-xyfucuorcfmqlopn) |
| 05:09.14 | dfgas-cr48 | dijib, if you enter wrong extension number is says that you have entered wrong extention and hangs up |
| 05:09.31 | dfgas-cr48 | if i enter anything other than 1-9 it will do the same thing |
| 05:21.17 | dijib | dfgas-cr48: it should have been fixed. did you do a dialplan reload? |
| 05:21.31 | dijib | maybe it was the same thing. do i still have creds? |
| 05:21.48 | dfgas-cr48 | no :( |
| 05:21.57 | dfgas-cr48 | i could load up team viewer |
| 05:22.17 | dijib | sure |
| 05:25.18 | dfgas-cr48 | password in pm |
| 05:27.25 | *** join/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
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| 07:23.29 | *** join/#asterisk doogienz (~doogie__@support.net24.co.nz) |
| 07:23.51 | doogienz | Hi to all - anyone see anything wrong with the syntax here? |
| 07:23.52 | doogienz | exten => s,n,ExecIf($["${CALLERID(num)}" = "anonymous" ]?Set(CDR(inboundsrc)="${SIP_HEADER(RealCallerID)}" |
| 07:24.06 | *** join/#asterisk topro (~quassel@host-62-245-142-50.customer.m-online.net) |
| 07:24.13 | doogienz | I keep on getting Set requires an '=' to be a valid assignment. |
| 07:26.04 | doogienz | Asterisk 1.8 |
| 07:29.06 | *** join/#asterisk cyborg-one (~cyborg-on@79-140-5-100.broadband.tenet.odessa.ua) |
| 07:31.59 | kaldemar | doogienz: there's a space after mous" and closing ('s missing for Set and ExecIf. |
| 07:32.01 | doogienz | Damnit - ignore - |
| 07:32.40 | doogienz | No it was the end - I'd forgotten the closing ) - it was only in a syntax highlighting editor I found it. |
| 07:32.46 | doogienz | exten => s,n,ExecIf($["${CALLERID(num)}" = "anonymous" ]?Set(CDR(inboundsrc)=${SIP_HEADER(RealCallerID)})) |
| 07:33.58 | doogienz | Soz - my bad - been a long day and had fun with P-Asserted-Identity for an upstream carrier. |
| 07:34.34 | dijib | can anybody tell me how many times you can take the dCAA? |
| 07:34.46 | dijib | and where do i find a study resource? |
| 07:37.45 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
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| 07:57.15 | dijib | https://www.google.com/url?q=http://www1.digium.com/en/training/asterisk/certifications/dcaa&sa=U&ei=xR2zUPDYGuOl2AXMuIHwBg&ved=0CAcQFjAA&client=internal-uds-cse&usg=AFQjCNHJ4J5q9zRPf1iQN71emjuS37Cpvw |
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| 08:13.19 | kaldemar | the dCAA seems to be fairly easy. |
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| 08:20.17 | R1ck | I have two asterisk servers on different locations, one in our office which has a Digium TE121 card for our ISDN lines and one in our datacenter location. I have configured a telephonenumber on the ISDN lines to be passed to the iax2 trunk to the asterisk in our datacenter. When I call the number with my mobile phone it works, my Polycom SIP phone connected to the asterisk in our datacenter is ringing, but the music on hold is terrible, inaudible, "choking"/stut |
| 08:22.17 | kaldemar | your message got cut at stut... |
| 08:23.48 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 08:24.11 | ChannelZ | stuttering |
| 08:43.46 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 08:52.06 | R1ck | oh |
| 08:52.14 | R1ck | [...] "choking"/stuttering.. Is this due to the connection between the two PBXes not having enough mbit upload from office to datacenter? |
| 08:54.39 | *** join/#asterisk timahvo1 (~rogue@197.181.221.253) |
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| 09:11.04 | kaldemar | R1ck: that's one possible reason. does the same occur on calls too or is it only music on hold? |
| 09:16.01 | *** join/#asterisk jaxon007_ (~jay@123.252.144.92) |
| 09:16.55 | jaxon007_ | Why Dahdi hangup delays 10 seconds to hangup? |
| 09:20.48 | *** join/#asterisk bombev (~bombev@PPPoE-Static-40-132.UnicsBG.Net) |
| 09:20.56 | bombev | hi all |
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| 09:45.12 | *** join/#asterisk Blue_Ice (~Blue_Ice@unaffiliated/blue-ice/x-2052838) |
| 09:45.31 | Blue_Ice | anyone over here with some mISDN experience? |
| 09:47.01 | *** join/#asterisk Coren (~kvirc@wikipedia/Coren) |
| 09:47.52 | Coren | Heya. Quick question: I can't seem to be able to change the channel language in lua (asterisk 11). Neither channel.language not channel.language() work (the latter states that language isn't a registered function). Any hints? |
| 09:50.50 | R1ck | kaldemar: no, calls are clear |
| 09:52.35 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 09:58.39 | wdoekes | Coren: channel.CHANNEL("language"):set("...") ? |
| 09:59.05 | Coren | wdoekes: ... that seems horribly conviluted. I'll go try. |
| 09:59.06 | con3x | Blue_Ice: I have a little |
| 09:59.13 | wdoekes | that's what extensions.lua tells me |
| 09:59.21 | wdoekes | -- channel.func_name(1,2,3):set("value") |
| 09:59.26 | wdoekes | -- channel["func_name(1,2,3)"]:set("value") |
| 09:59.27 | con3x | Blue_Ice: What are you trying to do |
| 09:59.31 | wdoekes | that last one is convoluted |
| 10:00.42 | Coren | wdoekes: That worked. MAN it's ugly. |
| 10:00.50 | wdoekes | Coren: you must remember that lua support is tacked on.. not an integral part |
| 10:01.29 | Coren | wdoekes: Yeah, I'm used to AEL; but the database support in lua is much cleaner. |
| 10:01.32 | wdoekes | the CHANNEL function has no way of telling the lua system that it's prettier if it's a language() function on the channel object |
| 10:02.01 | wdoekes | so.. when you want to say Set(CHE |
| 10:02.16 | wdoekes | so.. when you want to say Set(CHANNEL(xyz)=123) you'll have to do the aforementioned |
| 10:02.48 | Coren | Cleaner fix: |
| 10:02.52 | Coren | function lang() |
| 10:02.52 | Coren | <PROTECTED> |
| 10:02.53 | Coren | end |
| 10:02.56 | Coren | :-) |
| 10:03.08 | Coren | lang():set("en") |
| 10:03.39 | wdoekes | if it makes your code more readable, you should absolutely do that |
| 10:09.49 | ghost75 | i hate it when scammers let it ring only for couple of seconds and then drop call |
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| 10:43.08 | Blue_Ice | con3x: solved in the mean time, I couldn't make more than 2 outbound calls (while the group had multiple ports). Asterisk/mISDN didn't seem to try bringing up the L1 of the other ports. Solved short term by adding a misdn_check_l2l1 in the outbound dialplan. Will alter the "l1watcher_timeout" on next reboot to solve it module side |
| 10:43.29 | Blue_Ice | con3x: puzzled me since inbound calls above 2 were still working, but outbound not |
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| 10:52.49 | rolandow | goodmorning all! |
| 10:52.59 | rolandow | anybody using a siemens dect set here? |
| 10:55.26 | jaxon007_ | why DAHDI takes time to hangup call. I am using asterisk server as a PSTN gateway which has 4 port digium card installed along with 4 E1 lines. I have configured sip accounts in this machines and registering those SIP accounts from asterisk machines.. When I tried inbound call and when Extension on server is busy.. I got response in Asterisk server "BUSY". But Server takes 10 sec delay to hangup that call. What could be reason? |
| 10:55.48 | ghost75 | yes |
| 10:56.33 | *** join/#asterisk camerin (hoax@newelite.bshellz.net) |
| 10:59.02 | Chainsaw | rolandow: Yes, C450 and related handsets have been used here. |
| 11:10.21 | rolandow | Chainsaw: ok .. i hope you can help me :-) .. i am configuring a N300A base station .. i want it to use the fixed line by default.. but call extension numbers (as in 3 or 4 digits) through voip |
| 11:10.36 | rolandow | Chainsaw: the online help says I can use * or # in the phone number part, but it doesn't say how this works. |
| 11:10.54 | rolandow | Chainsaw: I tried both (### and ***), but it dials through the fixed line |
| 11:11.02 | rolandow | any suggestions? :) |
| 11:11.03 | WIMPy | Blue_Ice: Maybe you should upgrade? |
| 11:11.57 | rolandow | i already upgraded to the latest firmware |
| 11:12.06 | rolandow | oh .. that wasn't for me :) |
| 11:12.31 | WIMPy | jaxon007_: More info, please. |
| 11:12.36 | Chainsaw | rolandow: On a C450, you would hold down the green dial button for longer to override the routing decision. |
| 11:12.49 | Chainsaw | rolandow: There is a rudimentary dial plan, but I'm not convinced that this works. |
| 11:13.22 | rolandow | Chainsaw: hm.. i'd like to make it fool proof :) |
| 11:13.28 | rolandow | one of our employers is moving to france |
| 11:13.49 | rolandow | would be nice if the phone automatically chooses the right connection |
| 11:15.17 | Chainsaw | rolandow: I wouldn't trust the Siemens firmware for that; given the choice now I'd probably go for a more featureful DECT handset and put an ATA on the analog line to do the routing. |
| 11:15.55 | Chainsaw | rolandow: The Siemens UI is *very* slow, and I have found that the slowness (and inability to cancel a selection once made, particularly on dialling out) infuriates users. |
| 11:17.56 | *** part/#asterisk Coren (~kvirc@wikipedia/Coren) |
| 11:19.55 | jacekowski | Chainsaw: you haven't seen the worst of it |
| 11:20.15 | jacekowski | Chainsaw: i work with siemens PLCs and their new TIA software is slow on i7 with 8GB of ram and SSD |
| 11:20.30 | jacekowski | and i'm talking proper slow |
| 11:21.17 | WIMPy | Gigaset is not Siemens any more. |
| 11:21.46 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 11:21.48 | *** join/#asterisk ghost75 (~trechber@dslb-088-064-221-197.pools.arcor-ip.net) |
| 11:21.49 | jacekowski | when did that happen? |
| 11:21.55 | WIMPy | And I've been told there are interoperability issues since that. |
| 11:21.56 | ghost75 | (12:15:24) rolandow: would be nice if the phone automatically chooses the right connection <- would surprise me if it could do this |
| 11:22.19 | jacekowski | hmm |
| 11:22.28 | jacekowski | they are 20% siemens still |
| 11:22.31 | WIMPy | Not sure when that happened. But Gigaset even offer their own PBXs now. |
| 11:22.43 | jacekowski | well siemens always did offer PBXes |
| 11:23.14 | WIMPy | They still do. Either Siemens or SEN. |
| 11:24.17 | WIMPy | What's that about the right connection? |
| 11:25.20 | ghost75 | http://mks-technik.de/images/telefonanlagen/Hipath500.jpg |
| 11:26.26 | WIMPy | rolandow: YOu probably can't have overlapping numbers as you can in Asterisk. |
| 11:32.32 | rolandow | WIMPy: well i found out that * and # mean the literal # and # |
| 11:32.35 | rolandow | and * .. |
| 11:32.41 | rolandow | i thought they were wildcards |
| 11:32.52 | rolandow | but i have no clue yet what the R and P mean |
| 11:33.06 | rolandow | the manual says I can use 0-9,#,*,R,P in my dialplan |
| 11:33.36 | WIMPy | call back and pause? |
| 11:33.55 | rolandow | how would i use that in a dialplan? :) |
| 11:34.41 | WIMPy | Anyway, I'm pretty sure you can't do any routing based on the length. That's just not possible the way any normal telephony equipment works. |
| 11:35.29 | rolandow | hm.. that's too bad .. on my C470 i could suffix using the line number.. that doesn't seem to work either :( |
| 11:35.31 | WIMPy | You need distinct prefixes. |
| 11:37.01 | WIMPy | Dialplans shouldn't have ambiguities. |
| 11:38.29 | rolandow | yes.. but if i can't have a pattern using length .. then it's ambigious real quick |
| 11:38.41 | rolandow | i just need XXX and XXXX on the voip line :( |
| 11:38.48 | rolandow | stupid siemens |
| 11:40.13 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 11:45.01 | WIMPy | That's just not possible the way any normal telephony equipment works. |
| 11:45.26 | WIMPy | At the time the length is guessed the decisin will long have been taken. |
| 11:47.41 | WIMPy | And it could only ever work in countries with closed number plans anyway, as even 3 digit numbers may be valid phone numbers otherwise. |
| 11:47.55 | WIMPy | And that's not only emergency numbers. |
| 11:52.25 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.83.174) |
| 11:54.15 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 12:00.28 | *** join/#asterisk aurs (~aurs@110.84-49-69.nextgentel.com) |
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| 12:11.17 | *** join/#asterisk pbxman (c335d968@gateway/web/freenode/ip.195.53.217.104) |
| 12:11.21 | pbxman | hello |
| 12:15.19 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 12:16.21 | rolandow | that could be true, but at least it's up to me to decide then .. |
| 12:16.44 | rolandow | if asterisk supports it, why wouldn't a phone support it? |
| 12:19.05 | WIMPy | Maybe developers thought such a feature would be absurd? As has actually been the case before SIP made its way. |
| 12:21.29 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 12:30.13 | rolandow | but this is a SIP phone :) |
| 12:30.27 | rolandow | well, supporting SIP next to a fixed line. |
| 12:30.54 | rolandow | it would make more sense if you could define a dialing plan the same way as asterisk can .. |
| 12:33.38 | WIMPy | Asterisk is not the standard for Telephony. Hopefully. |
| 12:34.12 | Chainsaw | rolandow: It is a *very* limited UI. Consider other options, like an ATA on the analog line, and a better DECT phone. Your users will thank you. |
| 12:34.43 | WIMPy | How is going analog going to not make things worse? |
| 12:35.43 | WIMPy | Analog is 100% overlap only. |
| 12:37.55 | WIMPy | How long should it take for a SIP message to be retransmitted? I see retransmitted messages but no difference in the timestamp to the original message. |
| 12:40.53 | rolandow | Chainsaw: can't .. i already have the phone and the user has to take it with her to france .. |
| 12:41.17 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 12:41.26 | rolandow | Chainsaw: it's not a big deal, it's just a bit dissapointing :) if a fixed line is connected, i can have the user to choose for each outgoing call .. |
| 12:41.33 | Chainsaw | rolandow: Weigh the loss of face of having to purchase another phone against the eternal spite you will develop in the user going to france. |
| 12:41.56 | WIMPy | Doesn't france have an open dial pan? |
| 12:42.15 | WIMPy | plan |
| 12:42.24 | rolandow | WIMPy: what do you mean with an open dial plan? |
| 12:43.13 | rolandow | WIMPy: if i could just add XXX and XXXX to the dialplan to go to voip (except for 112), then i would be happy. i don't think phonenumbers in FR only have 4 digits |
| 12:43.19 | WIMPy | That valid numbers can have different lengths. |
| 12:43.20 | rolandow | all other numbers should go to the fixed line |
| 12:43.39 | rolandow | probably .. but i don't think they have the length of 3 or 4. |
| 12:43.43 | WIMPy | So that XXX or XXXX might be valid (external) numbers. |
| 12:44.12 | rolandow | even if they are .. *I* want them to go to voip .. if the user really wants to use the fixed line, then he could choose this line. |
| 12:44.58 | WIMPy | Seems like a bad hack to me. |
| 12:45.02 | rolandow | why? |
| 12:45.08 | rolandow | why is it bad if it works for me? |
| 12:45.34 | WIMPy | You are not the user. |
| 12:45.37 | ghost75 | what would be the easiest way to get cidname entries as csv? |
| 12:45.47 | rolandow | you can use any language to develop bad code |
| 12:45.50 | rolandow | doesn't mean the code is bad |
| 12:46.08 | rolandow | i am not the user .. but i am the one who is providing the phone to the user.. |
| 12:46.31 | rolandow | and i know how the user will be using the phone |
| 12:46.57 | rolandow | i don't really see why you're arguing that while you don't know the user and situation.. which i do |
| 12:47.02 | WIMPy | And the user won;t expect local calls to work? |
| 12:47.24 | rolandow | no, not to three digit numbers.. |
| 12:47.45 | rolandow | also you're defending a point which you are not sure about :) |
| 12:48.10 | WIMPy | But maybe you can see why someone who does phone firmware doesn;t hink that's asensible idea. |
| 12:48.10 | rolandow | france has a 10 digit numbering plan (unless the wiki page is incorrect) |
| 12:48.31 | *** join/#asterisk luckman212 (~luckman21@2001:470:8abb:0:211:32ff:fe10:cdc1) |
| 12:48.42 | ghost75 | are you still discussing siemens going over voip/normal phone line? |
| 12:48.46 | rolandow | yes.. i can see that.. but like i said: it would be nice if the user of the phone was able to make this choice |
| 12:48.51 | rolandow | instead of the firmware developer |
| 12:49.02 | rolandow | because the firmware developer isn't going to be the user for sure :) |
| 12:49.58 | rolandow | also the manual says the dial plan is made to use the cheapest route available .. |
| 12:50.00 | WIMPy | The user is always restricted by what developers put in to firmware. And that again is limited by the amount of time that can be invested. So things that don;t seem neccessary often won't make it. |
| 12:50.09 | rolandow | so i'd say you'll need more advanced patterns then just the prefix |
| 12:51.01 | WIMPy | Just prefix all internal numbers wit a * and you have a definite unique route. |
| 12:51.24 | rolandow | yes.. but in my opinion this limited dial plan is disappointing.. and if one would develop a phone that could also do sip calls, i don't see why the dial plan couldn't be a bit more advanced. |
| 12:51.48 | WIMPy | Pay for it. |
| 12:51.55 | Chainsaw | rolandow: I have been trying to explain to you that the SIP hybrid phone chosen is limited to the extreme. |
| 12:52.07 | ghost75 | <PROTECTED> |
| 12:52.27 | Chainsaw | rolandow: But you do not want to hear it. You want to have made the perfect choice. I am deeply sorry to have to report to you that no, you haven't. |
| 12:52.28 | rolandow | Chainsaw: yes, i agree .. am i not allowed to be a bit disappointed about that? :) |
| 12:52.38 | Chainsaw | rolandow: The phone is borderline unusable and your user will hate you if you do not address this now. |
| 12:52.51 | rolandow | Chainsaw: no, it's not a big issue, i already said that |
| 12:52.58 | WIMPy | You know how things work these days. If less than 90% of customers demand a feature, it won't happen. |
| 12:53.11 | rolandow | Chainsaw: i'm surprised that you know the user better than me.. |
| 12:53.57 | Chainsaw | rolandow: I maintain a phone system for 40 users spread across two offices. The average attention span to dial a phone number is 20 seconds. Anything that takes longer is doomed from the outset. |
| 12:54.06 | rolandow | WIMPy: yes, they even stripped some features that were available in a base station that i have at home for personal usage :( |
| 12:55.08 | Chainsaw | rolandow: So this system copes with users dialling without an area code, without a leading 9, etc. You can tell them off, but you must place the call. All within 20 seconds. Or it is "broken". |
| 12:55.15 | rolandow | Chainsaw: i maintain a phone system for 70 users spread across 7 locations |
| 12:56.09 | Chainsaw | rolandow: And the further removed you are from those locations, the less likely you are to get honest feedback. |
| 12:56.45 | rolandow | Chainsaw: this is a particular situation where the user decides to move abroad.. we want to help her with this phone.. i could easily say to get her own phone for the fixed landline. |
| 12:56.57 | Chainsaw | rolandow: But if you treat user feedback like you are treating the feedback here... I can see why the complaints have stopped. |
| 12:57.53 | rolandow | Chainsaw: i think you misjudge me (and my user) .. i mean, have we met before? :) |
| 12:58.10 | rolandow | have you ever figured that maybe your situation or users are different than mine? |
| 12:59.18 | Chainsaw | rolandow: Het is al goed. Doe het op jouw manier. |
| 12:59.24 | rolandow | WIMPy: i'm afraid you're right .. it's just a budget model indeed.. |
| 12:59.53 | rolandow | Chainsaw: everybody does it his own way .. at least i hope |
| 13:00.19 | Chainsaw | files rolandow under lost cause and awaits more fruitful cases |
| 13:00.41 | rolandow | right |
| 13:00.55 | rolandow | everybody who doesn't do it your way is a lost case |
| 13:01.05 | rolandow | or doesn't agree with you |
| 13:02.05 | rolandow | but well .. it's all in the name i suppose .. Chainsaw :) |
| 13:02.25 | Chainsaw | It is. Don't be a tree. |
| 13:02.52 | rolandow | i like trees :) |
| 13:03.48 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 16:02.24 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 16:02.24 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.0.1 (2012/11/05), 10.10.0 (2012/11/06), 1.8.18.0 (2012/11/06), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.13 (2012/10/09) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 16:05.38 | *** join/#asterisk andy09usa (~andy09usa@audotov.com) |
| 16:11.14 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 16:11.14 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:11.16 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 16:12.43 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
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| 16:21.04 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 16:23.00 | cusco | hey... |
| 16:23.17 | cusco | I'm having troubles performing outbound trough a sip provider... |
| 16:23.35 | cusco | it registers, sip show registry shows the peer as registered |
| 16:23.43 | cusco | but when dialing trough it: Failed to authenticate on INVITE |
| 16:24.17 | [TK]D-Fender | Your auth is suspect but it would help to actually SEE the call... |
| 16:24.53 | cusco | including sip debug? |
| 16:27.51 | cusco | [TK]D-Fender: http://ovh.tretas.eu/sip.txt |
| 16:28.20 | [TK]D-Fender | SIP, not core |
| 16:28.49 | *** join/#asterisk blee (~blee@70.118.107.77) |
| 16:29.23 | cusco | don't want the debug |
| 16:29.24 | cusco | ? |
| 16:29.27 | cusco | sip is also there |
| 16:30.07 | cusco | ok, refresh now The_Hatta |
| 16:30.08 | cusco | [TK]D-Fender: |
| 16:31.31 | cusco | it states proxy authentication required |
| 16:31.43 | cusco | also the relevant bit in sip.conf |
| 16:32.41 | cusco | is as follows: http://paste.debian.net/212486/ |
| 16:33.05 | ChannelZ | You're authing but it's rejecting it, which probably means it doesn't know who you are (it doesn't know your username) |
| 16:33.19 | [TK]D-Fender | [Nov 26 16:27:53] VERBOSE[16776] chan_sip.c: Reliably Transmitting (NAT) to 213.13.89.67:5070: |
| 16:33.26 | [TK]D-Fender | First, your provider should be "nat=no" |
| 16:33.30 | cusco | ok... |
| 16:34.01 | cusco | ChannelZ: I noticed : From: "asterisk" <sip:+351302037267@voip.sapo.pt>;tag=as157b7797 |
| 16:34.09 | cusco | should asterisk read somethingelse ? |
| 16:35.04 | [TK]D-Fender | cusco, that's a CID name based on the way you started your call. |
| 16:35.13 | [TK]D-Fender | I'd check the rest of your peer settings for this as well |
| 16:36.22 | cusco | all it states is username: +3513020XXXXX ; domain: voip.sapo.pt ; proxy: proxy.voip.sapo.pt ; proxy port: 5070 |
| 16:36.37 | cusco | so... I don't know what shall I need |
| 16:36.49 | cusco | weid is that I succesfully made calls in the past |
| 16:39.17 | cusco | the sip debug when registering has: From: <sip:+351302037267@voip.sapo.pt>;tag=as4f5ebbad |
| 16:39.25 | cusco | no callerid(name) |
| 16:39.33 | cusco | would that be a problem? |
| 16:45.54 | *** part/#asterisk orn (~orn@2a01:8280:101:0:c5:5d89:9496:9b3d) |
| 16:48.10 | p3nguin | That's a lot of stuff in the peer entry. |
| 16:49.17 | cusco | I agree !! |
| 16:49.48 | p3nguin | I would have started off more like this: http://paste.debian.net/212494/ |
| 16:50.09 | p3nguin | Much less stuff with less contradiction and less wrong parameters. |
| 16:50.48 | cusco | remotesecret |
| 16:50.48 | cusco | ok let me try those |
| 16:51.24 | p3nguin | remotesecret is used in cases where you have to authenticate on calls TO them, but they will never authenticate calls they send to you. |
| 16:51.55 | cusco | now instead I get [Nov 26 16:49:42] WARNING[7609]: chan_sip.c:20321 handle_response_invite: Received response: "Forbidden" from '"asterisk" <sip:asterisk@91.121.25.175>;tag=as1fa6be2b |
| 16:52.47 | p3nguin | That is probably because I took away the authuser. |
| 16:53.06 | p3nguin | authname, that is. |
| 16:53.49 | p3nguin | or maybe the fromdomain. |
| 16:54.04 | cusco | yes the fromdomain |
| 16:54.09 | p3nguin | Actually, that is probably it. |
| 16:54.10 | cusco | I added the fromuser and fromdomain |
| 16:54.12 | cusco | [Nov 26 16:51:49] NOTICE[7609]: chan_sip.c:20309 handle_response_invite: Failed to authenticate on INVITE to '"asterisk" <sip:+351302037267@voip.sapo.pt>;tag=as0eaf57a1' |
| 16:54.31 | cusco | it is back to previous state |
| 16:55.51 | cusco | any ideas? |
| 16:56.46 | cusco | I have bria on iphone and I think it works, if I can capture the sip call, would that help? |
| 16:57.08 | cusco | using the same credentials against voip.sapo.pt |
| 16:57.41 | *** join/#asterisk blee (~blee@70.118.107.77) |
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| 17:08.23 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:18.01 | *** join/#asterisk fiesch (~fiesch@pd95c43dc.dip0.t-ipconnect.de) |
| 17:20.08 | fiesch | hi all.. i have an issue with a ldap connected spa 504g which sends a sip string with whitespaces ("+49 89 XXXXXXX@[asterisk-ip]"). Is this something i need to fix in the spa or can i fix this within * ? |
| 17:20.11 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:20.11 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:21.09 | fiesch | (number with whitespaces comes from customers' corporate dir ) |
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| 17:28.23 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 17:28.53 | [TK]D-Fender | fiesch, It's your dialplan. You can fix spaces there, but I'd recommend fixing it on the device itself |
| 17:35.28 | *** join/#asterisk parasitodelsur (~wtf@c-98-200-53-71.hsd1.tx.comcast.net) |
| 17:35.36 | parasitodelsur | waz up guys |
| 17:35.43 | parasitodelsur | is a done deal. |
| 17:35.50 | parasitodelsur | I moved to business class. |
| 17:35.54 | parasitodelsur | yay! |
| 17:36.43 | p3nguin | parasitodelsur: This is Comcast Account Management. We'll be shipping your new SMC high speed internet gateway out to you right away. Thank you for your business. |
| 17:36.57 | parasitodelsur | p3nguin: LOL |
| 17:37.07 | parasitodelsur | by the way they did agree to install on bridge mode |
| 17:37.22 | p3nguin | I wouldn't have used anything but my own modem. |
| 17:37.23 | parasitodelsur | I got it on writing and part of my contract |
| 17:37.30 | parasitodelsur | they dont allow it |
| 17:37.33 | parasitodelsur | :( |
| 17:37.36 | parasitodelsur | I tryed |
| 17:37.43 | p3nguin | But bridge mode will be better than another layer of NAT. |
| 17:37.51 | parasitodelsur | yeap. |
| 17:37.55 | p3nguin | At least you'll be able to use your own router and not their crap. |
| 17:38.02 | parasitodelsur | exactly |
| 17:38.14 | *** join/#asterisk paulc (~root@unaffiliated/paulc) |
| 17:38.25 | parasitodelsur | p3nguin: msg |
| 17:38.43 | p3nguin | Try it. |
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| 18:28.54 | ghost75 | now it took me ages to store and read asterisk db to/from csv |
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| 18:34.30 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 18:38.45 | p3nguin | It shouldn't have been that difficult. |
| 18:40.47 | chris_n | any idea why a polycom 331 on a different subnet from the * server would not update to DST automagically? Everything else works fine with it. |
| 18:41.43 | [TK]D-Fender | chris_n, TZ detection isn't "magic"\ |
| 18:41.52 | [TK]D-Fender | chris_n, Assuming it is would be a mistake |
| 18:42.23 | chris_n | it was said tongue-in-cheek; nothing is magic the last time I checked :) |
| 18:43.23 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
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| 18:57.40 | ghost75 | (19:39:15) p3nguin: It shouldn't have been that difficult. <- with AMI and perl ? |
| 18:58.14 | p3nguin | Just because you choose to use tools that make it more difficult for you to do it does not mean it is that difficult to do. |
| 18:59.37 | ghost75 | then how you store cidnames without making it difficult |
| 19:00.34 | p3nguin | I store all my caller id stuff in CDR and in the asterisk DB. |
| 19:01.21 | ghost75 | store in cdr ? |
| 19:02.55 | p3nguin | Anything related to caller id will be found in my CDR and in my astDB. |
| 19:04.07 | ghost75 | u store callerid as name in cdr ? |
| 19:04.14 | ghost75 | not as number? |
| 19:05.00 | p3nguin | You aren't making sense to me. I've already told you two times, anything pertaining to caller id will be found in my CDR and my AstDB. |
| 19:06.08 | ghost75 | anyway i meant like restoring cidnames in case of disaster or whatever |
| 19:06.09 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 19:07.59 | p3nguin | For the fourth time, it's in the AstDB. |
| 19:08.21 | ghost75 | you dont get what i want to say |
| 19:08.57 | p3nguin | If my AstDB would get blown up, AND if all my backups of it would get blown up, I don't care if I don't have caller ID names stored anymore. I'll just query the numbers when the people call again and store it again. |
| 19:09.25 | ghost75 | ok |
| 19:11.35 | *** join/#asterisk kontinuity (~Adium@122.178.254.121) |
| 19:20.28 | dijib | p3nguin: what digium certs do you hold? |
| 19:20.48 | dijib | and how many times can i take the dcaa |
| 19:21.06 | p3nguin | You should only need to take it one time. |
| 19:21.22 | dijib | some parts i have not touched |
| 19:21.22 | dijib | pri |
| 19:21.38 | p3nguin | If you can't pass in one try, you should have read The Book once. |
| 19:21.48 | dijib | the thing that looked like DAHDI but was iXXXXX or something |
| 19:22.02 | p3nguin | dundi? |
| 19:22.22 | dijib | not dahdi.. |
| 19:22.34 | dijib | pbx company in australia. |
| 19:22.38 | p3nguin | oh |
| 19:22.43 | dijib | wheres my testking damnit? |
| 19:22.52 | dijib | maybe that 1 |
| 19:23.36 | dijib | Asterisk 1.8.18.0 built by root @ swissarms on a x86_64 running Linux on 2012-11-25 06:59:03 UTC |
| 19:23.47 | dijib | drunk. |
| 19:24.55 | _Corey_ | Taking certification tests whilst drunk is an unusual approach |
| 19:25.32 | dijib | its all about the Designated Driver. |
| 19:27.11 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 19:29.13 | file | holds zero certifications |
| 19:34.31 | *** join/#asterisk AkkerKid (~AkkerKid@50-200-18-202-static.hfc.comcastbusiness.net) |
| 19:35.45 | AkkerKid | YO! Good {Insert your respective local time period here} Everyone! |
| 19:37.22 | AkkerKid | My newest project is to figure out how to initiate a robocall to a customer when there purchase is ready for pickup. I have available to me a database that would show that information and a decent grasp of dialplan programming. What's my next step? |
| 19:37.41 | AkkerKid | their purchase* |
| 19:38.11 | dijib | hardware? |
| 19:38.24 | dijib | oh |
| 19:38.26 | dijib | nvmd |
| 19:39.28 | dijib | i would think you would need to create a callfile when the pickup is ready. |
| 19:39.36 | p3nguin | You can use the AMI or basic shell scripting to initiate the phone call. |
| 19:39.44 | _Corey_ | AkkerKid: Assuming you have some Asterisk knowledge, I did a how-to session at the 2011 Astricon on AMD and robo-calling... there's a video somewhere |
| 19:40.07 | p3nguin | I have a small section of dial plan that plays some predetermined sound files to a callee. |
| 19:40.19 | AkkerKid | I'd love to fire off a croned script that checks the DB for new completions and calls the customers and plays a recorded message. |
| 19:40.34 | *** join/#asterisk sjobeck (~sjobeck@70-89-188-5-or.portland.hfc.comcastbusiness.net) |
| 19:41.54 | sjobeck | hey, hi, all, question for you. Any one know how to ring an extension somehow (a Bogen pager) but not answer the call. The reason is that I still need to ring another extension. So, this would need to start a separate second call. some how. thoughts? |
| 19:42.13 | _Corey_ | AkkerKid: last one on the bottom right... http://www.astricon.net/2011/videos/videopresentations.aspx |
| 19:43.07 | jacekowski | hmm, any sip phones that can do ipv6? |
| 19:43.07 | AkkerKid | sweet |
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| 19:43.22 | sjobeck | _corey_ was that to me? |
| 19:44.01 | jacekowski | sjobeck: so you want call to one extension to ring two phones? |
| 19:44.18 | jacekowski | sjobeck: take a look at how paging is done |
| 19:44.57 | sjobeck | yes, but, one of those phones can not "answer", so to speak, as that will stop the ringing. to be clear, we are using "paging" already to the Bogen. |
| 19:45.18 | sjobeck | i will be ringing a loud bell in the factory when the phone on the wall is ringing |
| 19:45.36 | sjobeck | i wish i could connect the bell to the phone but thats not possible |
| 19:46.39 | _Corey_ | sjobeck: Was what to you? |
| 19:47.03 | sjobeck | that URL to astriCON ? |
| 19:48.18 | _Corey_ | no, it was directed at AkkerKid |
| 19:49.06 | sjobeck | thx :) |
| 19:50.04 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 19:50.20 | sjobeck | so …… any one know how I can send some audio file of some type to the Bogen, triggered by my call, without it answering my other call that is ringing the phone on the wall? (my hurdle here is that the Bogen goes off hook instantly & I can't have that stop the ringing call to the handset) |
| 19:50.54 | sjobeck | (and, dang it, that I can't connect the handset directly to the Bogen, wish i could, problem solved, but cant) |
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| 19:55.29 | _Corey_ | sjobeck: Why do you need to have a phone ringing on a wall somewhere in order to play an audio file into your paging system? |
| 19:55.56 | _Corey_ | the relationship between these two things isn't clear |
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| 20:01.24 | sjobeck | good Q. i don't. i need to ring the Bogen amp when the handset is ringing in this loud factory. but if i simply ring the extension not the Bogen, that answers the call. and the call dies. |
| 20:03.18 | [sr] | i'm tired |
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| 20:12.40 | sjobeck | _corey_ thoughts? |
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| 20:17.11 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 20:21.45 | _Corey_ | sjobeck: Maybe if you explain step-by-step what you're trying to accomplish... It's still unclear why you're ringing a phone and your paging system together |
| 20:32.27 | sjobeck | thx. ok, here goes. call comes in to the factory. rings the factory phone on the wall. can't connect handset to Bogen. Want Bogen to ring when handset rings. Want to place a second call to the Bogen. Want that second call to not kill the first call. How to trigger/spawn a second independent call? That second call would be answered, which is fine, but that wouldn't kill the first call. The second call would play a au |
| 20:33.49 | [TK]D-Fender | sjobeck, "core show application originate" |
| 20:34.02 | [TK]D-Fender | sjobeck, Or Call Files. Or CLI Originate. Or AMI Originate |
| 20:35.55 | sjobeck | ok, so, let me follow you, you are saying to have the call come in from the PSTN, hit some program, that program would originate 2 calls, one to x123 (handset) & another to x124 (Bogen). The call to x123 would be answered by human. the call to x124 would time out on its own after 20 seconds. |
| 20:38.36 | _Corey_ | sjobeck: I had to read it a couple times, but if I understand this correctly, you're actually trying to play a ringing noise into the PA system while the phone rings also? |
| 20:38.47 | sjobeck | exactly |
| 20:39.06 | sjobeck | (the turkey has me writing poorly today i guess ) |
| 20:40.09 | _Corey_ | OK, so obviously there are a lot of ways to do that but it sounds like you're looking for a low-tech approach |
| 20:42.00 | _Corey_ | so, I'd suggest you look at the call spool sample file in the Asterisk sources... fork something that copies a call spool file you've created into /var/spool/asterisk/outgoing. That call should just connect your PA device to an extension that plays a ringing sound and hangs up |
| 20:47.14 | sjobeck | thats great. i will. we have used call files forever. can you suggest a specific way to fork to that second call? |
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| 20:49.35 | *** join/#asterisk krotos (~d3v1l@host116-28-dynamic.56-82-r.retail.telecomitalia.it) |
| 20:49.39 | krotos | hi all |
| 20:52.08 | krotos | i've got to ring a mobile phone trought a trunk ( calling out to this trunk) and just the mob.phone is ringing i've got to hangup. |
| 20:52.23 | krotos | is a good idea to use Asterisk / Manager |
| 20:52.41 | krotos | or an agi script? |
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| 20:54.45 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 20:55.15 | _Corey_ | sjobeck: Many ways... the crudest is to use System() to execute a cp |
| 20:56.16 | *** part/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 20:56.50 | sjobeck | _corey_ reading on that now |
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| 21:01.22 | _Corey_ | sjobeck: I'm also a fan of the Originate app for something like this, but use whatever you're comfortable with |
| 21:02.54 | p3nguin | dijib: I can't stop laughing over that. |
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| 21:10.07 | ghost75 | is it now common in us to fry a turkey ? |
| 21:11.50 | [TK]D-Fender | It's common in the US to fry EVERYTHING. |
| 21:12.11 | ghost75 | lol |
| 21:12.18 | [TK]D-Fender | Case in point : "friend butter" <- Google at your own risk |
| 21:12.24 | [TK]D-Fender | Fried* |
| 21:12.27 | [TK]D-Fender | gah |
| 21:12.47 | ghost75 | fried butterfinger |
| 21:15.46 | p3nguin | fried beer |
| 21:16.21 | p3nguin | So... in case you missed it... |
| 21:16.38 | p3nguin | dijib thinks DUNDi is a PBX company in Australia. |
| 21:16.45 | ghost75 | is this real, fried butter? |
| 21:17.15 | p3nguin | OH NOSE! What is going to happen to deep fried Twinkies now that Hostess is no more? |
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| 21:35.40 | AkkerKid | turkey fried in peanut oil is SOOO FREAKING GOOD! |
| 21:36.05 | AkkerKid | I was considering putting up with my family and driving 250 miles to eat it this past holiday... |
| 21:36.24 | drmessano | I prefer peanuts fried in turkey oil |
| 21:36.26 | drmessano | So good |
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| 21:36.36 | AkkerKid | _Corey_ Still alive? |
| 21:38.20 | _Corey_ | AkkerKid: Sure, what's up? |
| 21:39.15 | AkkerKid | How would I initiate an outbound call from a shell script and implement AMD? Or is that not how I should do it? |
| 21:42.15 | AkkerKid | can't just... asterisk -rx "Originate(SIP/number,exten,number)" right? |
| 21:42.32 | p3nguin | Create an extension that utilizes AMD(). Then you'll originate the call using either shell script or AMI. The call will consist of two parts: the phone and the extension you created which uses AMD(). |
| 21:43.08 | p3nguin | Usage1: channel originate <tech/data> application <appname> [appdata] |
| 21:43.11 | p3nguin | Usage2: channel originate <tech/data> extension [exten@][context] |
| 21:43.25 | p3nguin | Originate() is a dial plan app and does not run with asterisk -rx. |
| 21:44.56 | AkkerKid | but channel originate does work with asterisk -rx? |
| 21:45.02 | p3nguin | Yes. |
| 21:45.06 | AkkerKid | sweet |
| 21:45.29 | p3nguin | channel originate is a CLI command. asterisk -rx runs the CLI and executes commands that follow. |
| 21:46.26 | AkkerKid | ah got it./ |
| 21:47.14 | _Corey_ | AkkerKid: I'm a little distracted at the moment, but you're getting good advice |
| 21:48.21 | AkkerKid | good either way |
| 21:48.31 | p3nguin | For something very similar to what you are trying to do, I create one new context with two extensions in it. One extension is 's' which contains my routine with AMD(), WaitForSilence(), and Playback()... the other extension is a pattern matching all possible phone numbers I will be calling (in this case, _NXXNXXXXXX). |
| 21:48.50 | AkkerKid | what's the purpose of the second? |
| 21:49.34 | p3nguin | The extension for the phone numbers does things like marking things in the database, setting caller ID number, etc. (things that need to be done when the call is sent out). |
| 21:49.58 | p3nguin | Oh, and Dial()ing the number, of course. |
| 21:50.40 | p3nguin | The s extension is where everything happens once the call is answered. |
| 21:50.49 | AkkerKid | oh i see. |
| 21:50.55 | pabelanger | anybody else having issues with toronto.voip.ms today? |
| 21:52.30 | p3nguin | The shell script then runs (asterisk -rx "channel originate Local/$NumToCall@special_context extension s@special_context" &) |
| 21:53.19 | p3nguin | NumToCall, in my case, is pulled from a list of numbers. For you, your API will have to feed the number from your shipping system. |
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| 21:54.33 | p3nguin | My script is a little more complicated, because I check the time of day before running the originate. Don't need to make robocalls too early or too late in the day. |
| 21:56.54 | p3nguin | If you whip up something, I'd take a look at it if you'll pastebin it. |
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| 22:23.17 | SeRi | wow what a pita. I need to cancel my residential account.... I thought that them been part of comcast they could also take care of that. |
| 22:26.55 | drmessano | Nope |
| 22:27.13 | p3nguin | They probably don't have any idea that you don't still want to keep your residential service. |
| 22:27.24 | drmessano | The residental and business parts of Comcast are isolated |
| 22:27.31 | drmessano | They can't seem to get any of it correct |
| 22:32.42 | SeRi | lol |
| 22:33.09 | SeRi | well I was told to hold off from the disconnect so I can overlap it |
| 22:33.19 | *** join/#asterisk timahvo1 (~rogue@197.181.221.253) |
| 22:33.51 | SeRi | I just fought off a 50 dollar charge for a tech to come fix the addressable tab. |
| 22:34.41 | SeRi | wtf is wrong with them. |
| 22:35.05 | dijib | yo |
| 22:35.19 | drmessano | I need to have them disconnect my business class and give me residential back |
| 22:35.23 | dijib | come back to conf SeRi |
| 22:35.56 | dijib | can you not join? |
| 22:35.56 | SeRi | dijib: I am there |
| 22:36.04 | SeRi | where you at? |
| 22:36.08 | dijib | coming |
| 22:36.09 | SeRi | drmessano: why? |
| 22:36.13 | SeRi | Is cheaper |
| 22:36.16 | dijib | some reason my dialplan is a little wonky |
| 22:36.50 | p3nguin | They wanted to charge you to fix THEIR equipment on THEIR system outside of your premises? |
| 22:36.53 | SeRi | drmessano: 5 IP cidr block telephone and internet for 100 dollars a month |
| 22:37.00 | SeRi | p3nguin: Yes. |
| 22:37.14 | p3nguin | As you put it, WTF is wrong with them? |
| 22:37.27 | SeRi | lol |
| 22:37.44 | dijib | fixing dialplan bug |
| 22:37.55 | p3nguin | seri: So... dijib thinks that DUNDi is a PBX company in Australia. |
| 22:37.57 | SeRi | dijib: originate while you fix it you lazy |
| 22:38.06 | SeRi | ROFL! |
| 22:38.07 | SeRi | No way! |
| 22:38.11 | p3nguin | Way. |
| 22:38.16 | SeRi | That was one of the questions in dcaa |
| 22:38.19 | p3nguin | I know. |
| 22:38.24 | p3nguin | That's where he got it from. |
| 22:38.25 | _Corey_ | p3nguin: Didn't he say he was drunk? |
| 22:38.28 | dijib | it wont let me into the conference |
| 22:38.41 | SeRi | chan originate dijib |
| 22:38.49 | SeRi | _Corey_: excuses |
| 22:39.20 | p3nguin | _corey_: I think he said it, but I don't know if I believed it. I also will never be able to forget how funny that is. |
| 22:39.22 | _Corey_ | SeRi: I found it noteworthy because he was also trying to take one of Digium's online certification tests |
| 22:39.49 | p3nguin | I figured that was the only place it would show up. |
| 22:40.24 | p3nguin | Crocodile DUNDi |
| 22:40.29 | SeRi | _Corey_: ah. I am with p3nguin though.... |
| 22:40.32 | SeRi | lol |
| 22:41.05 | p3nguin | What was your score on the test? |
| 22:41.24 | SeRi | 92 |
| 22:41.37 | p3nguin | I didn't do so good. I got a 97%. |
| 22:41.51 | SeRi | I had a hard time with pri stuff |
| 22:41.56 | p3nguin | I wish I could see what two I missed. |
| 22:43.16 | SeRi | Meh I pass... I am going for the dcap as soon as next year. |
| 22:43.22 | SeRi | work will pay for the crash course |
| 22:43.31 | p3nguin | I probably won't ever take the dcap. |
| 22:43.38 | p3nguin | It's not really necessary for me. |
| 22:43.59 | SeRi | I am doing it because work wants to send me to training |
| 22:44.08 | SeRi | why not... |
| 22:45.12 | p3nguin | If I already know the material and I am capable of completing the exam successfully, how will I benefit from actually doing it? |
| 22:46.29 | SeRi | p3nguin: From jobs. A lot of my customer ask for specialization and this is the way the company shoes that we specialize |
| 22:46.45 | SeRi | s/shoes/shows/ |
| 22:47.22 | SeRi | They also want me to go for RHCE 6 and RHCA |
| 22:47.37 | SeRi | I have RHCT/RHSA/RHCE on 5 |
| 22:47.45 | SeRi | well T became SA. |
| 22:48.48 | SeRi | I am not sure I have the time for A. Is one hell of a test. |
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| 23:03.42 | dijib | p3nguin: did u end up taking the dcaa? |
| 23:04.20 | p3nguin | (1641.37) <p3nguin> I didn't do so good. I got a 97%. |
| 23:05.14 | SeRi | That should been a 100 |
| 23:05.32 | SeRi | :P |
| 23:06.34 | p3nguin | I know. I'll go kill myself now. |
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| 23:07.41 | dfgas-cr48 | ughhh |
| 23:07.46 | *** join/#asterisk angryuser_laptop (~angryuser@2a02-8422-1230-bb00-0597-5e4b-890b-2b99.rev.sfr.net) |
| 23:07.53 | SeRi | p3nguin: LOL |
| 23:07.55 | dfgas-cr48 | dijib, yo |
| 23:08.32 | dfgas-cr48 | i fixed one issue last night |
| 23:09.21 | *** join/#asterisk artyx (U2FsdGVkX1@junction.googleplex.net) |
| 23:09.44 | dfgas-cr48 | its weird but might make sense to the gurus, i removed your blacklist stuff from inbound and now i can enter more than 1-9 |
| 23:10.12 | SeRi | ? |
| 23:10.18 | dfgas-cr48 | however if you enter wrong number it tells you that you have dialed a wrong extension then hangs up on you |
| 23:10.21 | SeRi | That does not make sense |
| 23:10.42 | SeRi | The first part. |
| 23:10.56 | [TK]D-Fender | p3nguin: http://www.quickmeme.com/meme/3rxgqg/ |
| 23:11.26 | SeRi | [TK]D-Fender: LOL |
| 23:11.36 | dfgas-cr48 | SeRi, yah idk either, if you dial 1-9 it would dial what ever extension. but anything over that would tell you that this number is not in service and hang up on you like you were calling with a black listed number |
| 23:12.43 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 23:12.49 | dfgas-cr48 | but if i removed the blacklist part of my inbound everything would work fine, it won't replay choices though |
| 23:14.45 | SeRi | somebody did it wrong. |
| 23:14.51 | SeRi | dijib: you around? |
| 23:15.22 | p3nguin | I worked on it once and verified it. That's where you should have left it. |
| 23:15.41 | p3nguin | "If it ain't broke, break it." |
| 23:17.20 | SeRi | word |
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| 23:22.31 | artyx | There is an option/var =ATTENDED_TRANSFER_COMPLETE_SOUND ... Is there any documentation on the parameters of the file it uses |
| 23:30.23 | dfgas-cr48 | p3nguin, the inbound was not changed at all |
| 23:30.30 | dfgas-cr48 | since you verified it |
| 23:30.41 | p3nguin | What part is broken? |
| 23:30.58 | dfgas-cr48 | just the inbound part of the dialplan |
| 23:31.37 | SeRi | dfgas-cr48: did you do a core verbose and see where it was failing? |
| 23:32.19 | dfgas-cr48 | dijib, thought it was because it said it was calling 11@voipms-inbound |
| 23:32.37 | dfgas-cr48 | but when you dial 1 it does 1@voipms-inbound |
| 23:32.43 | dfgas-cr48 | so that can't be it |
| 23:33.42 | dfgas-cr48 | i know dijib did the core verbose |
| 23:34.21 | p3nguin | That transaction did not make sense whatsoever. |
| 23:34.27 | p3nguin | (1730.23) <dfgas-cr48> p3nguin, the inbound was not changed at all |
| 23:34.32 | p3nguin | (1730.41) <p3nguin> What part is broken? |
| 23:34.32 | p3nguin | (1730.58) <dfgas-cr48> just the inbound part of the dialplan |
| 23:34.40 | p3nguin | Does. Not. Compute. |
| 23:35.18 | dfgas-cr48 | right now it will call any of my extensions, which is what i want, however i have noticed now that if you dial the wrong extension it will tell you that you dialed the wrong extension then hang up on you |
| 23:35.40 | p3nguin | What did you want it to do if you enter the WRONG extension? |
| 23:36.20 | dfgas-cr48 | p3nguin, you said that I should have not messed with the dialplan, and i am saying inbound was not changed, but inbound is the issue, |
| 23:36.30 | p3nguin | Then it was changed. |
| 23:37.03 | dfgas-cr48 | replay menu and and let you choose again |
| 23:37.22 | p3nguin | Look at your extension 'i' in that context. |
| 23:37.40 | dfgas-cr48 | hang on] |
| 23:44.33 | dfgas-cr48 | p3nguin, http://pastebin.com/Qz6QxJXx right now the blacklist is commented out. if you uncomment it out then it will only allow me to enter 1-9. if you enter anything else then it will tell you the number it not in service or what ever the blacklist message is then beeps real fast like busy tone |
| 23:45.08 | p3nguin | "fast busy" is a congestion tone. |
| 23:45.37 | p3nguin | This is all messed up and doesn't even make sense. |
| 23:45.57 | p3nguin | Why would there be a verbose of "the call reached h" right in the middle of your DID extension? |
| 23:46.30 | p3nguin | And you don't have the i extension like I mentioned. |
| 23:46.52 | p3nguin | i = invalid, when you enter digits in WaitExten() and BackGround(). |
| 23:48.09 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 23:48.10 | SeRi | the whole thing is/was brokwn |
| 23:48.16 | p3nguin | That Goto() immediately after your IVR file plays also doesn't make sense. |
| 23:48.19 | p3nguin | It wasn't broken when I left it. |
| 23:48.25 | p3nguin | Now it's all fucked up. |
| 23:48.37 | p3nguin | Someone had to touch it. |
| 23:48.43 | dfgas-cr48 | oh i put that in last night to see if it would fix |
| 23:48.43 | p3nguin | Couldn't just leave it alone. |
| 23:49.07 | p3nguin | Fix up the things I told you about and repaste. |
| 23:49.59 | p3nguin | And if your extension actually is 920319XXXX, you forgot the underscore on the front of it. |
| 23:50.09 | p3nguin | 920319XXXX isn't valid, _920319XXXX is. |
| 23:50.59 | [TK]D-Fender | same => n,Goto(t,1); <- you are forcing a timeout right after calling backgroud. You are shooting yourself in the foot |
| 23:51.05 | [TK]D-Fender | This is a pretty bad AA design |
| 23:51.06 | dfgas-cr48 | no its just my phone number, i was unsure why it is like that, in the samples i have seen it showed something different |
| 23:51.07 | p3nguin | You also don't have a 'goodbye' label, but you have a GotoIf() that uses it. |
| 23:51.39 | p3nguin | The design works fine when someone hasn't fucked it up. |
| 23:51.49 | [TK]D-Fender | include => internal; <- we don't see what's in here so hard to say what's valid |
| 23:51.55 | p3nguin | It actually works great; I use it every day. |
| 23:52.01 | [TK]D-Fender | You should also stop ending every line with a ";" |
| 23:52.10 | p3nguin | Why? You think that's breaking the dial plan? |
| 23:52.21 | p3nguin | ('cause it isn't) |
| 23:52.39 | [TK]D-Fender | line 12 = broken |
| 23:53.01 | p3nguin | That's not the only one that's broken. I've told him at least four things that need fixed. |
| 23:53.03 | [TK]D-Fender | same => n,BackGround(IVR); <-- starts backgrounding. |
| 23:53.13 | [TK]D-Fender | same => n,Goto(t,1); <- jumps immediately to FAIL |
| 23:53.21 | p3nguin | covered it. |
| 23:53.37 | dfgas-cr48 | SeRi, you were right :D |
| 23:53.39 | [TK]D-Fender | same => n,WaitExten(8); ; Return here on invalid <- you RETURN to waitexten? |
| 23:54.00 | [TK]D-Fender | p3nguin: There a more recent and hopefull better version around now? |
| 23:54.20 | p3nguin | I'm still waiting on him to fix the four or five things I told him about and repaste. |
| 23:54.35 | [TK]D-Fender | Line 17 looks like a total waste |
| 23:54.56 | p3nguin | I'm kind of pissed off that I personally corrected and verified the thing, but now it's all changed and fucked up. |
| 23:55.56 | [TK]D-Fender | exten => f,1,Goto(fax-in,fax,1); <- WTF? |
| 23:56.37 | [TK]D-Fender | same => n,Goto(voipms-inbound,9203198188,1); <- so much for MASKING this up top in the pattern |
| 23:56.38 | p3nguin | It's supposed to run through the IVR, wait for exten, timeout, play the IVR again, wait for exten, timeout, play the IVR again, timeout, say sorry you're having problems, run exten 't' which says goodbye and hangs up. |
| 23:56.49 | [TK]D-Fender | Also a bad way to do this... you are RESETTING your COUNT <- |
| 23:57.11 | p3nguin | Sorry you're having problems could be moved to t without too much problem. |
| 23:57.11 | [TK]D-Fender | This sample is a ClusterFuck (tm) |
| 23:57.56 | p3nguin | I'm slightly interested in knowing why it was changed after it was working properly. |
| 23:58.04 | dijib | uhmmm..... |
| 23:58.47 | dijib | http://pastebin.com/bzk2AUW9 |
| 23:58.57 | dfgas-cr48 | p3nguin, http://pastebin.com/1dv8LP4r this is the one you said was good. this is the one i started with that 1-9 works fine, if you dial 10 or anything else it come up as blacklisted |
| 23:59.18 | dfgas-cr48 | this is what started the changes of trying to figure out what was wrong |
| 23:59.41 | [TK]D-Fender | Idgthere is no 1-9 in there |
| 23:59.48 | [TK]D-Fender | dfgas-cr48: there is no 1-9 in there |
