| 00:00.56 | gusto | what is PIC? |
| 00:04.11 | gusto | ah |
| 00:07.54 | qakhan | [TK]D-Fender u there? |
| 00:08.02 | qakhan | mu life saver :) |
| 00:08.06 | qakhan | me* |
| 00:08.09 | qakhan | my* |
| 00:08.45 | [TK]D-Fender | <PROTECTED> |
| 00:08.50 | [TK]D-Fender | it just GAVE you the parameter to use... |
| 00:09.33 | qakhan | u mean i run make -fPIC |
| 00:11.12 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 00:12.30 | qakhan | [root@asterisk asterisk-11.1.0-rc1]# ./configure -fPIC |
| 00:12.31 | qakhan | configure: error: unrecognized option: -fPIC |
| 00:12.31 | qakhan | Try `./configure --help' for more information. |
| 00:16.55 | WIMPy | Looks like the Asteisk configure doesn't have that option. |
| 00:17.41 | qakhan | i m using asterisk 11.1.0-rc1 |
| 00:20.06 | Snivets | Anyone in the room have any idea why some Snom M9 handsets not connected to a repeater (at least as far as I know, 2 base stations, 6 phones) would be periodically emitting beeps, around 1x per 3min? |
| 00:21.44 | Snivets | I just tried the progressinband=no |
| 00:27.06 | apb1963 | Snivets I had the same problem! Only I don't have a Snom and it turned out to be IRC that was beeping. >:-0 |
| 00:27.36 | apb1963 | Drove me nuts for 4 days |
| 00:27.46 | apb1963 | I almost called the cops |
| 00:28.05 | apb1963 | That would of been embarressing..... hey... the beeping is coming from YOUR house! |
| 00:44.56 | *** join/#asterisk kaushal (~kaushal@182.71.248.194) |
| 00:45.00 | kaushal | Hi |
| 00:45.31 | kaushal | Can i set pager duty application using asterisk with time conditions mapped to Nagios Alerts? |
| 01:03.00 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 01:04.53 | Snivets | lol |
| 01:05.08 | Snivets | well, I sure hope it's user error. |
| 01:07.09 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 01:07.16 | SeRi | guys |
| 01:10.32 | slav3_kitten | http://www.dailymail.co.uk/news/article-2239686/Sweet-Treats-snaps-Rogue-Scouts-limb-female-charity-arm-wrestling-contest.html |
| 01:11.35 | slav3_kitten | what's up SeRi |
| 01:12.26 | SeRi | going through hell |
| 01:13.14 | SeRi | 5060 is been block |
| 01:13.22 | SeRi | comcast swares they are not |
| 01:15.40 | SeRi | can any body help me with a remote udp scann? |
| 01:17.27 | kikohnl | SeRi what can I do to help? |
| 01:18.14 | SeRi | kikohnl: can you msg me? |
| 01:18.31 | *** join/#asterisk deo (~deo@203.177.214.75) |
| 01:20.03 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 01:22.13 | apb1963 | no don't message him.... other people are having the same problem |
| 01:22.39 | *** join/#asterisk ice_strike (~androirc@94-192-112-241.zone6.bethere.co.uk) |
| 01:22.51 | SeRi | apb1963: ??? |
| 01:23.00 | apb1963 | ???? |
| 01:23.25 | SeRi | dijib: conf |
| 01:23.37 | SeRi | apb1963: who is having the same issu? |
| 01:23.43 | apb1963 | me |
| 01:24.06 | apb1963 | ISP is blocking my users packets |
| 01:24.18 | SeRi | who is your isp? |
| 01:24.29 | ice_strike | which asterisk book do you recommendq to learn about caller center dialer or predictive dialer |
| 01:24.32 | apb1963 | not my ISP... his ISP |
| 01:25.01 | apb1963 | which is in Bangladesh |
| 01:25.14 | ice_strike | I like to write predictive dialer script |
| 01:25.46 | SeRi | ~book |
| 01:25.46 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 01:28.05 | *** join/#asterisk lanning (~lanning@50-193-22-25-static.hfc.comcastbusiness.net) |
| 01:28.09 | ice_strike | is that good to learn about agent dialer design? |
| 01:28.39 | slav3_kitten | isn't 4th edition coming out soon SeRi? |
| 01:29.12 | SeRi | MOTHER FU((*&&&****(!!!!!! |
| 01:29.27 | SeRi | I change port 5080 and it works |
| 01:29.33 | SeRi | they are filetering the damn port. |
| 01:29.36 | SeRi | bastardos! |
| 01:35.50 | [TK]D-Fender | ice_strikewhich asterisk book do you recommendq to learn about caller center dialer or predictive dialer <- I cannot imagine anyone actually writing a book on how to write a dialer. |
| 01:36.02 | [TK]D-Fender | ice_strike: It's too obvious a concept and with too many examples |
| 01:36.05 | apb1963 | SeRi: You're lucky. mine doesn't work no matter what port I use. |
| 01:36.13 | [TK]D-Fender | ice_strike: And those who write the, SELL them |
| 01:36.17 | [TK]D-Fender | them* |
| 01:43.52 | dijib | ahhh comcast you blow |
| 01:44.24 | dijib | apb1963: your on comcast business? |
| 01:45.40 | apb1963 | no |
| 01:46.02 | ice_strike | ok I see tkd |
| 01:46.07 | dijib | what country? |
| 01:46.20 | apb1963 | As I said.... <apb1963> not my ISP... his ISP <apb1963> which is in Bangladesh |
| 01:46.32 | ice_strike | which asterisk book do you recommend |
| 01:46.49 | dijib | ~book |
| 01:46.49 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 01:47.17 | ice_strike | and any more? |
| 01:47.20 | lanning | I moved from comcast residential to business to get out of the filters. (could not use any other DNS servers than theirs) |
| 01:48.00 | lanning | and to get a static IP |
| 01:49.32 | qakhan | is there any list of applicationa and functions in asterisk 10.10.0 |
| 02:04.35 | *** join/#asterisk blee (~blee@67.8.206.215) |
| 02:06.00 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 02:06.46 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 02:06.46 | SeRi | dijib: |
| 02:06.55 | SeRi | dijib: can you hear me? |
| 02:08.24 | SeRi | slav3_kitten: you around? |
| 02:15.22 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 02:22.40 | SeRi | am online? |
| 02:23.29 | SeRi | damn |
| 02:25.16 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 02:25.48 | *** join/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
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| 02:43.32 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 02:50.17 | SeRi | anybody available to do a test? |
| 02:56.17 | kikohnl | Sure |
| 02:57.10 | *** part/#asterisk deo (~deo@203.177.214.75) |
| 02:59.18 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 03:00.19 | SeRi | p3nguin: are you available for a test? |
| 03:00.26 | *** join/#asterisk Smirker (~x@124-170-105-181.dyn.iinet.net.au) |
| 03:15.43 | dijib | SeRi: in my conf |
| 03:18.49 | *** join/#asterisk dfgas-cr48 (~user@71-90-33-37.dhcp.ftbg.wi.charter.com) |
| 03:47.44 | dijib | SeRi: im back in there |
| 03:47.52 | dijib | if you chan redirect |
| 03:47.55 | dijib | that call |
| 03:48.39 | SeRi | dijib: one sec |
| 03:53.06 | SeRi | yeap.... defently the line |
| 03:56.23 | SeRi | dijib: hey |
| 04:29.21 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-uzshbtbanrsomvqk) |
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| 05:08.36 | *** join/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com) |
| 05:10.03 | nny | anyone have suggestions for a simple HA setup? Open for suggestions, just some way to have two servers act out redundancy in a failure. |
| 05:25.18 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 05:34.37 | *** join/#asterisk Aethrs (~secret@static-71-251-98-122.tampfl.fios.verizon.net) |
| 05:35.02 | Aethrs | I'm trying to compile asterisk 1.8.18.0 under Debian, but make menuconfig won't allow me to use chan sip. I'm missing some package(s) I imagine, any guesses which one(s)? |
| 05:38.02 | Maliuta | Aethrs: install the source package and use it's rules file as a guide |
| 05:38.39 | Maliuta | Aethrs: and if I were you I'd be building that into a package anyway, otherwise what's the point of using a package managed system? |
| 05:44.22 | *** join/#asterisk elico (~Thunderbi@109.64.187.2) |
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| 06:05.42 | dfgas | ughhh |
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| 06:59.51 | ChannelZ | Aethrs: openssl-dev |
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| 07:41.42 | *** join/#asterisk salz212 (~chatzilla@203.215.176.22) |
| 07:51.07 | salz212 | Hi, is there any documentation available for rtt (round trip time) unit for asterisk versions? I need to know the difference for rtpqos,audio,all for all versions. Is it stable yet even in Asterisk 11? |
| 07:52.36 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
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| 08:00.37 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 08:00.37 | *** mode/#asterisk [+o blitzrage] by ChanServ |
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| 08:41.14 | Rico29 | hi all |
| 08:41.31 | Rico29 | is this problem resolved in latest versions of asterisk ? (1.8) |
| 08:41.31 | Rico29 | http://support.freepbx.org/forum/freepbx/users/asterisk-become-mad-when-a-dns-problem-occur |
| 08:42.45 | WIMPy | 1/8 is far from latest. |
| 08:42.48 | WIMPy | 1.8 |
| 08:43.35 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.190) |
| 08:43.55 | WIMPy | And that looks as if you might want to enable dnsmgr. I think that was available since 1.8. |
| 08:48.25 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 09:01.06 | bombev | hi all |
| 09:01.19 | WIMPy | lo you |
| 09:05.39 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:3d11:bd45:72ac:b3e7) |
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| 09:10.02 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 09:10.39 | v0lZy | hi WIMPy |
| 09:12.13 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
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| 09:38.46 | ChannelZ | meh. |
| 09:56.58 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 09:58.45 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
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| 10:00.52 | *** join/#asterisk salz212 (~chatzilla@203.215.176.22) |
| 10:02.18 | *** join/#asterisk `paul (70c9b82a@gateway/web/freenode/ip.112.201.184.42) |
| 10:03.26 | `paul | i have an IVR that transfer calls to another asterisk server. is there a way to know how many ongoing/concuurent calls are happening to that server so if i reach a limit say 10 i would transfer to my other asterisk server? |
| 10:03.46 | salz212 | guys I am curious about RTCp stats from Asterisk.. Need to confirm one thing i.e. units of RTT round rtip time |
| 10:04.31 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:04.48 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:04.56 | salz212 | it is in milliseconds or micro seconds? or it can be both |
| 10:06.48 | WIMPy | `paul: core show function group<tab> |
| 10:07.39 | `paul | thanks |
| 10:09.09 | kaldemar | salz212: what are you looking at? |
| 10:09.18 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 10:09.39 | *** part/#asterisk deo (~deo@222.127.13.226) |
| 10:09.50 | salz212 | channel variable: RTP qos audio all... "ssrc=962343404;themssrc=4248;lp=0;rxjitter=0.000333;rxcount=679;txjitter=0.000000;txcount=882;rlp=0;rtt=37781.548000" . I guess it is too small for paste bin. |
| 10:11.01 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 10:11.02 | salz212 | here rtt value seems to be in micro .. but I have seen it coming in 0.037 format as well. whats the difference.. is this version specific\ |
| 10:14.39 | salz212 | any idea? |
| 10:18.56 | kaldemar | messy. if you enable rtcp debug in CLI there is (sec) following the figure. the channel variable does not have a unit. |
| 10:20.16 | salz212 | okay thanks |
| 10:20.20 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 10:30.27 | kaldemar | salz212: looks like it is milliseconds in rtcp debug and seconds in the channel variable. |
| 10:30.41 | salz212 | which version are you using? |
| 10:30.47 | kaldemar | so the (sec) in rtcp debug is wrong anyway. |
| 10:30.55 | kaldemar | i took a look at 11.0.1 source. |
| 10:34.28 | *** join/#asterisk chris- (~chris@p5DD185CB.dip0.t-ipconnect.de) |
| 10:34.31 | chris- | hi |
| 10:34.35 | chris- | :) |
| 10:35.07 | chris- | i have a question about ami over http. Is there any documentation for all the commands? |
| 10:41.16 | plundra | You could do listcommands? :) |
| 10:41.39 | chris- | sure, but it would be nice to get something about, where I need which parameters etc |
| 10:42.00 | chris- | thats why i ask, would be faster if its already exists instead of trying everything :) |
| 10:42.27 | Chainsaw | chris-: manager show commands |
| 10:42.30 | Chainsaw | chris-: On the Asterisk console. |
| 10:43.30 | chris- | yes but for example login needs 2 parameters, I knew it cause of an example, but I wanted to know if there are more than this one which need parameters |
| 10:46.36 | plundra | chris-: The wiki seems to have a good list of them. |
| 10:46.44 | plundra | https://wiki.asterisk.org/wiki/display/AST/AMI+Actions |
| 10:47.47 | plundra | The doxygen-docs should probably have them too I suppose :-) Can't imagine that list being written by hand. |
| 10:49.55 | chris- | thanks, seems great :) |
| 10:50.23 | chris- | btw, I already was programming with an API for AMI. What can I do with the actionid? |
| 10:51.15 | chris- | is it interesting for me if i'm programming a GUI for AMI? |
| 10:54.21 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.188) |
| 10:54.44 | plundra | I believe it's for keeping track of which action a perticular response is for. |
| 10:55.01 | chris- | ah ok |
| 10:55.06 | chris- | thx |
| 10:55.49 | plundra | I don't use it my self. Not sure if you can do it via the xml-thing, can you? |
| 10:56.40 | chris- | I didn't use the http interface but I wanted to know which possibilities I have with AMI |
| 10:56.54 | chris- | the other API was for C and didn't use http |
| 10:57.18 | chris- | but I had the possibility to use actionids, too |
| 10:57.56 | chris- | I just put a empty string and it worked, but I didn't know for what I could use it |
| 10:58.18 | chris- | just asked for it cause I saw it again at this documentation |
| 11:12.27 | WIMPy | chris-: It's just a "your reference" style field. |
| 11:13.10 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
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| 12:08.59 | chris- | ok thx |
| 12:10.08 | *** join/#asterisk surferboy (~surferboy@gatekeeper.isoft.co.za) |
| 12:10.19 | surferboy | I'm a total asterisk noob |
| 12:10.32 | surferboy | when I dial a direct extension I get the voice mail prompts |
| 12:10.48 | surferboy | if I get transfered from another extension I don't get the voice mail prompts |
| 12:10.49 | surferboy | why? |
| 12:11.14 | *** join/#asterisk blee (~blee@70.118.107.77) |
| 12:11.33 | kaldemar | surferboy: look at CLI with verbosity enabled when you do those. |
| 12:11.36 | plundra | Different contexts? |
| 12:12.59 | surferboy | way to much info |
| 12:13.13 | kaldemar | ~pb |
| 12:13.13 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 12:13.17 | surferboy | kaldemar, what should I look for in the confs |
| 12:13.25 | surferboy | lol |
| 12:13.56 | kaldemar | surferboy: you should start by looking at what happens. if you cannot interpret what you see, use pastebin and let others look at it. |
| 12:14.12 | kaldemar | when the difference is found, then is the time to play with configs. |
| 12:14.17 | surferboy | is there like a grep for that |
| 12:14.34 | kaldemar | no. put it all there. |
| 12:15.28 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 12:15.57 | kaldemar | salz212: what are you doing? |
| 12:17.13 | salz212 | tried to contact you.. guess didnt work as expected. Anyways.. about 11.0.1 RTCP code.. i am getting rtt around 30,000 |
| 12:18.12 | kaldemar | salz212: 1. don't contact people with dcc like that. 2. you're using a private address when initiating dcc. |
| 12:21.18 | salz212 | I do not intend to contact you in private, so chill. |
| 12:22.55 | salz212 | Just wanted to know which file yo looked at to check the units. thats it. |
| 12:24.38 | kaldemar | next time just ask here instead of poking me with a useless dcc. res/res_rtp_asterisk.c |
| 12:26.40 | *** join/#asterisk leifmadsen (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:26.40 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:30.10 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-58-4.user.veloxzone.com.br) |
| 12:31.06 | anonymouz666 | I was testing ConfBridge yesterday... got dissapointed because I didn't read about the lack of MCU support :( |
| 12:33.17 | WIMPy | MCU? |
| 12:34.17 | Chainsaw | Manic Customer Unmuter. |
| 12:34.46 | Chainsaw | If there is prolonged use of a raised voice on a muted channel, the mute comes off. |
| 12:36.21 | anonymouz666 | WIMPy: Multi-Conference Unit |
| 12:36.38 | Chainsaw | anonymouz666: I like mine better. |
| 12:37.02 | WIMPy | What does that mean? |
| 12:37.24 | Chainsaw | WIMPy: It looks to be a confbridge-in-a-box. From Cisco. |
| 12:37.44 | Chainsaw | WIMPy: And to keep echo cancellation viable, I wouldn't want to daisy-chain confbridges. |
| 12:37.56 | kaldemar | he probably means the lack of transcoding/scaling. |
| 12:38.05 | WIMPy | No. That's probably a bad idea. |
| 12:38.21 | WIMPy | And the reason it is disabled in the PSTN. |
| 12:38.26 | anonymouz666 | it does mean you can see everybody in a videoconf. you don't need to just follow the talker. |
| 12:38.32 | anonymouz666 | or last marked |
| 12:38.35 | anonymouz666 | or first_marked |
| 12:38.50 | anonymouz666 | or trigger a DTMF to be the talker |
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| 12:43.44 | *** join/#asterisk deo (~deo@112.198.90.172) |
| 12:43.50 | *** part/#asterisk deo (~deo@112.198.90.172) |
| 12:45.19 | *** join/#asterisk Tim_Toady (~moi@178.128.20.156.dsl.dyn.forthnet.gr) |
| 12:46.49 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 12:51.15 | *** join/#asterisk nextime (~nextime@unaffiliated/nextime) |
| 12:51.18 | nextime | hello all |
| 12:51.23 | nextime | i'm experiencing a strange issue |
| 12:51.39 | nextime | i have a grandstream ht 503 connected to my pstn and registered to * |
| 12:52.04 | nextime | if i call from another sip phone registered to my * all is working |
| 12:52.25 | nextime | if i receive a call from the pstn and i redirect the call with a Dial() to an internal phone all is working |
| 12:52.46 | nextime | but if i try to do an originate cli command with SIP/pstn ( the account of the ht503 ) |
| 12:52.55 | nextime | asterisk freeze for few minutes |
| 12:53.07 | nextime | even in debug i don't see anything strange |
| 12:53.16 | nextime | it just refuse the call with a permission denied |
| 12:53.33 | nextime | any idea on how to solve that or what can i check? |
| 12:57.05 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 12:57.08 | Ice_Strike | Hi |
| 12:58.41 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 13:12.45 | *** join/#asterisk Venaran (~venaran@72-53-156-125.cpe.distributel.net) |
| 13:14.11 | *** join/#asterisk rgsteele (~rgsteele@12.150.6.65) |
| 13:16.53 | *** join/#asterisk Ashly-Juniperbri (~chatzilla@217.36.31.1) |
| 13:17.14 | *** join/#asterisk deo (~deo@112.198.90.172) |
| 13:17.38 | Ashly-Juniperbri | Afternoon all! |
| 13:18.19 | *** join/#asterisk [TK]D-Fender (~TK]D-Fend@216-191-106-165.dedicated.allstream.net) |
| 13:18.57 | p3nguin | nextime: core set verbose 3, then run the originate. Pastebin everything you see come out on the CLI. |
| 13:23.21 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
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| 13:23.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 13:26.36 | *** part/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 13:28.26 | chris- | shouldnt we use verbose 5? (just a question from another noob :) ) |
| 13:28.56 | p3nguin | Why 5? |
| 13:29.33 | p3nguin | At level 4, dnsmgr crap gets added... which clutters the output. |
| 13:29.49 | chris- | i read in books that lvl 5 should be set |
| 13:30.09 | p3nguin | Nothing pertaining to the extensions running gets more verbose above 3. |
| 13:30.19 | chris- | ok |
| 13:30.24 | chris- | thx for the information :) |
| 13:30.34 | p3nguin | So 3 is the ideal level for basic call debugging. |
| 13:30.42 | *** join/#asterisk slav3_kitten (~kitten@unaffiliated/slav3-kitten/x-0866809) |
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| 13:31.09 | p3nguin | I believe level 7 is where CDR executions are added, so if you are debuggin CDR, go at least that high. |
| 13:31.19 | chris- | ok |
| 13:31.33 | chris- | i dont know what cdr means, so i think i dont need it *g |
| 13:31.38 | p3nguin | ~cdr |
| 13:31.38 | infobot | from memory, cdr is Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also - it could also be Compact Disc Recordable, see cdrw |
| 13:32.01 | chris- | ah |
| 13:32.02 | chris- | thx |
| 13:32.17 | p3nguin | You probably need it, but maybe not as much as other people need it. |
| 13:32.24 | chris- | btw bridge between channels means, they can communicate together? |
| 13:32.29 | p3nguin | Yes. |
| 13:32.34 | chris- | ok |
| 13:32.47 | p3nguin | Bridging is "connecting" multiple channels together. |
| 13:33.02 | chris- | i just need to work 1 semester |
| 13:33.05 | chris- | im a student^ |
| 13:33.26 | chris- | should do some things with asterisk, never worked with voip before |
| 13:33.29 | p3nguin | In a regular call, it is usually just two channels being bridged. In a conference, you could get several hundred channnels being bridged into one "call." |
| 13:33.37 | chris- | but its interesting :) |
| 13:34.58 | chris- | and thx for help |
| 13:35.02 | p3nguin | Now I'm wondering if I can manually bridge two channels which are already bridged together with two other channels that are also bridged together. |
| 13:35.22 | chris- | i'm just looking through the commands for AMI |
| 13:35.26 | p3nguin | Allowing for four endpoints on a single "call." |
| 13:35.32 | chris- | https://wiki.asterisk.org/wiki/display/AST/ManagerAction_Bridge |
| 13:35.52 | chris- | here it's just about two channels |
| 13:36.36 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.82.63) |
| 13:38.08 | Ice_Strike | When I answer the call, it hang up automatically. |
| 13:38.15 | Ice_Strike | This what it show on the CLI |
| 13:38.17 | Ice_Strike | <PROTECTED> |
| 13:38.24 | Ice_Strike | then it get disconnected |
| 13:39.21 | Venaran | thats been replaced has it not |
| 13:39.24 | Ice_Strike | Ah This function is depricated. :/ |
| 13:41.01 | Ice_Strike | How do I find out which version supported SetCDRUserField |
| 13:43.50 | kaldemar | where did you come up with that app? |
| 13:44.53 | kaldemar | the app was removed in 1.6.0. |
| 13:45.05 | [TK]D-Fender | Ice_Strike, That app you are using is making configs for an OLD version of Asterisk. That APP does not exist anymore |
| 13:45.28 | [TK]D-Fender | Ice_Strike, Expect it to cause more problems like this all around |
| 13:47.13 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 13:47.46 | anonymouz666 | this app is old |
| 13:48.01 | anonymouz666 | use CDR function |
| 13:48.21 | *** join/#asterisk tm1000 (tm1000@2600:3c01::f03c:91ff:fe93:fa45) |
| 13:51.12 | chris- | someone already worked with AMI over HTTP here? |
| 13:52.32 | *** part/#asterisk deo (~deo@112.198.90.172) |
| 13:52.47 | WIMPy | What's the idea of using http? |
| 13:53.18 | chris- | I just want to see what is all possible with AMI |
| 13:53.26 | chris- | or better say I need to do this:) |
| 13:53.28 | [TK]D-Fender | chris-, : |
| 13:53.30 | [TK]D-Fender | ~book |
| 13:53.30 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 13:53.31 | [TK]D-Fender | ^^^ |
| 13:53.40 | *** join/#asterisk kuruption (kuruption@cocaine.addikt.org) |
| 13:53.52 | chris- | :) |
| 13:54.02 | [TK]D-Fender | chris-, And Digium's WIKI. They all list the functions, for which HTTP doesn't offer any different than the usual socket |
| 13:54.09 | *** join/#asterisk Venaran (~venaran@2001:470:b16a::110) |
| 13:54.47 | chris- | ok |
| 13:54.49 | chris- | :) |
| 13:54.52 | chris- | thx |
| 13:55.47 | *** join/#asterisk serafie (~erin@nat/digium/x-fuutwdhiqqczqagx) |
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| 14:00.02 | [TK]D-Fender | <chris-> here it's just about two channels <- it is. Because you hook B onto A. And then another time to hook C ont A, and again for D onto A, until you have gang-piled channels ONE AT A A TIME into a katamari or horrific proporion. |
| 14:00.25 | chris- | ^ |
| 14:00.27 | chris- | ok |
| 14:00.28 | chris- | :) |
| 14:00.41 | chris- | so if i bridge two channels i get a new channel? |
| 14:00.45 | [TK]D-Fender | no |
| 14:00.50 | [TK]D-Fender | they simply bridge |
| 14:01.01 | chris- | i just didn't test it cause i didn't setup enough clients |
| 14:01.53 | [TK]D-Fender | You are bridging 2 channels, nothing more. They are hooked. There doesn't suddenly become more of them. |
| 14:02.04 | chris- | ok |
| 14:02.22 | slav3_kitten | [TK]D-Fender, now confbridge creates a mux channel right? |
| 14:02.25 | chris- | and if i want to bridge 3 channels? should i bridge the first ones and then one of these with the third? |
| 14:02.30 | chris- | or both of them? |
| 14:02.38 | *** join/#asterisk qakhan (~qakhan@208.253.91.58) |
| 14:02.41 | *** join/#asterisk keycruncher (~Adium@oh-67-76-244-251.sta.embarqhsd.net) |
| 14:04.55 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:09.04 | [TK]D-Fender | slav3_kitten, Don't know, never touched it. |
| 14:09.17 | slav3_kitten | *nods* i'm googling my ass off |
| 14:09.31 | [TK]D-Fender | chris-, Ask yourself when and why you would be using AMI for this in the first place.... |
| 14:09.39 | slav3_kitten | but i'd think chris- would want like a conference call setup instead of bridging like 3+ channels |
| 14:09.50 | Ice_Strike | is there any asterisk files I could what version it is without doing asterisk -v ? |
| 14:10.04 | Ice_Strike | i could find out* |
| 14:11.14 | Ice_Strike | maybe I could hex /usr/sbin/asterisk to find out what the version is |
| 14:12.15 | Chainsaw | Ice_Strike: The strings command does end up showing the version string eventually, yes. |
| 14:12.22 | [TK]D-Fender | Ice_Strike, Your config files could be based on ancient crap (they are) and not prove anything |
| 14:12.41 | *** join/#asterisk bchia (~Adium@nat/digium/x-daizsdfcorkkqcrz) |
| 14:12.43 | [TK]D-Fender | Ice_Strike, Just connect to CLI |
| 14:19.24 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:22.47 | *** join/#asterisk chris_n (~Chris@184.7.21.42) |
| 14:26.55 | chris- | [TK]D-Fender i just got it as a task to do ;-) like I said im a student who needs to work for 1 semester at a firm |
| 14:27.30 | chris- | looking for which are the possibilities with AMI |
| 14:27.31 | [TK]D-Fender | chris-, Got what "task" to do? You are asking piecemeal questions without a clear picture of something you're actually trying to accomplish... |
| 14:28.35 | chris- | and testing them |
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| 14:50.18 | *** join/#asterisk PbxMan (c335d968@gateway/web/freenode/ip.195.53.217.104) |
| 14:50.20 | PbxMan | hello |
| 14:51.20 | Venaran | good morning |
| 14:52.25 | bombev | Good afternoon I would say :) |
| 14:55.04 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.174) |
| 14:56.46 | LiuYan1 | good night zzz~ |
| 14:57.48 | *** part/#asterisk LiuYan1 (~LiuYan@222.125.131.151) |
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| 15:09.30 | SeRi | g/m all |
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| 15:19.00 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:19.01 | blizzow | When outside callers are connected to my asterisk system, they hear a weird ticking sound every second or so. People on my end do not hear the sound at all. My dahdi timing seems up to snuff (showing mostly 99.98% and 99.99%). My redfone tdmoe box shows my PRIs in OK condition. My PRI provider says our lines are clean. I'm using 1.8.12 on a dell poweredge 710 with 8GB ram. There is almost no load on the server. Where should I start looking at what co |
| 15:19.38 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
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| 15:22.43 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:31.42 | AkkerKid | YAR! |
| 15:32.20 | dfgas | PIRATES!!! |
| 15:34.48 | *** join/#asterisk leifmadsen (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
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| 15:40.02 | adeeln | anyone happen to have a mysql query that calculates the number concurrent calls using the cdr's that they wouldn't mind sharing? |
| 15:42.33 | *** join/#asterisk spditner (~simon@206-248-134-127.dsl.teksavvy.com) |
| 15:44.12 | spditner | in extensions.conf.sample, [stdexten] will cause the destination extension of any calls that end in noanswer/busy to be recorded in cdr's as stdtexten-noanswer/busy... |
| 15:44.25 | spditner | so what's the best practice for recording the original extension dialled? |
| 15:44.45 | p3nguin | seri: How is your packet loss today? |
| 15:44.56 | p3nguin | Derroche de los paquetes? |
| 15:45.40 | p3nguin | seri: The best practice is to NEVER use the sample files for production systems. |
| 15:45.43 | p3nguin | err... |
| 15:45.49 | p3nguin | spditner: The best practice is to NEVER use the sample files for production systems. |
| 15:46.42 | spditner | I realise that, I'm only referencing the example so as to have an example around which to discuss |
| 15:47.16 | p3nguin | But not using that model will rectify the problem. |
| 15:48.01 | spditner | what model are you suggesting, jumping to (tag)'s instead of extensions? |
| 15:48.25 | p3nguin | Skip all the macro crap. It isn't necessary. |
| 15:48.34 | spditner | it uses gosub |
| 15:48.48 | p3nguin | Oh? I guess I need to see a more updated sample file. |
| 15:49.02 | spditner | macro doesn't alter the cdr |
| 15:49.11 | spditner | but it's been deprecated in favour of gosub |
| 15:49.18 | spditner | however gosub does alter the cdr |
| 15:49.38 | *** join/#asterisk fritz09 (~Adium@2002:542e:50e1:e472:6d92:28d5:332c:500e) |
| 15:50.44 | SeRi | p3nguin: is bad. very bad. |
| 15:50.53 | SeRi | my upload is 60Kbps |
| 15:51.04 | p3nguin | That's screaming fast! |
| 15:51.12 | SeRi | rofl!!!!!! |
| 15:51.21 | p3nguin | Are you supposed to have like 5 meg up? |
| 15:51.29 | SeRi | two techs are coming in today. |
| 15:51.35 | SeRi | 10 meg |
| 15:51.39 | SeRi | lol |
| 15:52.05 | WIMPy | SeRi: That's like hte situation I hade when I had KDG. |
| 15:52.31 | SeRi | WIMPy: what was the issue? |
| 15:52.43 | SeRi | My issue is the addressable tap and the line coming in out side |
| 15:52.48 | WIMPy | throttling |
| 15:52.55 | p3nguin | Here's what goes on behind the doors at Comcast. "Hey Bill, don't you think it will be funny to tell this guy he'll get 10 Mbps upload but throttle him to just 60 kbps?" |
| 15:53.15 | SeRi | they are suppose to run a RG11 and fix the tap. |
| 15:53.20 | SeRi | p3nguin: LOL |
| 15:53.34 | p3nguin | What speed did you get with your residential modem? |
| 15:53.44 | SeRi | 23/4 |
| 15:53.51 | SeRi | now is 16/10 |
| 15:54.01 | p3nguin | You just said it was not 10, though. |
| 15:54.15 | SeRi | o thats because the line is all jacked up |
| 15:54.28 | p3nguin | I'm asking what you were actually seeing before the change to business. |
| 15:54.33 | SeRi | there is issues on the line and the ta as I explain before |
| 15:54.37 | SeRi | o |
| 15:54.38 | SeRi | ok |
| 15:54.51 | p3nguin | Were you getting all 4 Mbps ? |
| 15:54.57 | SeRi | nope |
| 15:55.02 | p3nguin | Also 60 k? |
| 15:55.07 | SeRi | roughly around 1 meg at most |
| 15:55.11 | SeRi | usually 768 kbps |
| 15:55.23 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:55.23 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:55.29 | p3nguin | I would expect to see the exact same speed with another modem. |
| 15:55.43 | p3nguin | The line quality doesn't degrade just by changing the CPE. |
| 15:56.02 | SeRi | Indeed. BUT. I am about 100% sure the modem is fucked up too |
| 15:56.15 | SeRi | the tech did not let the modem finish provisioning |
| 15:56.18 | p3nguin | We knew that before they gave it to you. |
| 15:56.22 | SeRi | than it took like 3hrs to fix |
| 15:56.42 | p3nguin | It takes about 1 second for the modem to download the cfg. |
| 15:56.54 | SeRi | He finally got the modem to download the bin file |
| 15:57.17 | SeRi | yeap the tech was a dumb ass and created a cluster fuck here |
| 15:57.39 | gusto | cool |
| 15:57.46 | gusto | that's like everywhere :-D LOL |
| 15:57.49 | p3nguin | Are they using dynamic configs on business class? |
| 15:57.55 | SeRi | he had no clue what was going on. |
| 15:58.14 | SeRi | all he cared was to finish fast so he can go home |
| 15:58.26 | SeRi | he evn told me |
| 15:58.43 | SeRi | "shit now I am going to be stuck on traffic" |
| 15:58.47 | SeRi | p3nguin: Not sure. |
| 15:59.16 | SeRi | p3nguin: The tech's manager is coming to my house today |
| 15:59.17 | p3nguin | I usually don't have to raise my voice to the people at the cable company, but I recently had to yell at a few people. |
| 16:00.02 | SeRi | I wouldnt doubt it |
| 16:00.17 | p3nguin | Stupid people trying to charge a reconnection fee to press a few keys on the keyboard... I had enough of that crap. |
| 16:00.48 | SeRi | God do I hate that. |
| 16:01.21 | p3nguin | I can understand if the customer had a PHYSICAL disconnection, where a person has to actually drive out and connect some cables, but when it is a soft disconnect... GRRRRRR! |
| 16:01.48 | *** join/#asterisk ruied (~AndChat66@bl9-234-165.dsl.telepac.pt) |
| 16:01.56 | p3nguin | "Just push the fucking button and I'll reset the device." |
| 16:02.18 | p3nguin | That'll be $60. Please come again. |
| 16:02.53 | SeRi | lol |
| 16:03.29 | p3nguin | Screw them. |
| 16:03.59 | p3nguin | If the guy comes to your house and still can't understand the problem, put him on the phone to me. |
| 16:04.19 | p3nguin | I'll make him wish he never met Comcast. |
| 16:04.40 | SeRi | LOL |
| 16:04.44 | SeRi | I will do that! |
| 16:04.55 | SeRi | be on the watch of an incommin channel to your conf |
| 16:05.07 | p3nguin | The bad part about phone calls is that people can hang up. :( |
| 16:05.14 | SeRi | shit just sit and listen while I put up the show on speaker |
| 16:05.26 | p3nguin | Then I can yell from the speaker phone. |
| 16:05.38 | SeRi | lmao |
| 16:06.56 | SeRi | current test results 7720 Kbps/62 Kbps |
| 16:07.08 | p3nguin | Wow that's bad. |
| 16:10.26 | p3nguin | I remember way back in the day when we didn't even have internet service from the cable co. |
| 16:10.53 | p3nguin | Once I heard they were deploying it, I hooked up my modem/gateway device and just waited. |
| 16:11.17 | p3nguin | One night around 12 midnight, I saw the sync light stop blinking. |
| 16:11.40 | p3nguin | I switched my route to use the cable gateway and it was online! I was so happy. |
| 16:11.56 | SeRi | lol nice |
| 16:12.00 | p3nguin | I did a speed test, and I had 64kbps symmetric. |
| 16:12.09 | p3nguin | 64k up, 64k down. |
| 16:13.00 | SeRi | man I remember those days. |
| 16:13.03 | p3nguin | I waited and waited for them to officially begin providing internet service, which was about two weeks, and then I called to ask them to subscribe me to high speed internet. |
| 16:13.45 | p3nguin | So for that long I got dial-up speeds, but it freed up my phone line. |
| 16:13.51 | p3nguin | I was happy with that at the time. |
| 16:15.39 | *** join/#asterisk jkroon (~jkroon@dsl-244-30-18.telkomadsl.co.za) |
| 16:15.47 | SeRi | no kidding man. |
| 16:16.01 | SeRi | I was happy with 1/768 back in the days |
| 16:16.09 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 16:16.13 | SeRi | that was my first speed tier with tw. |
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| 16:45.08 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 16:49.37 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.82.63) |
| 16:50.07 | p3nguin | tedt |
| 16:51.20 | p3nguin | Anyone here use android IRC and understands the autocorrect setting? I have the "disable autocorrect" setting turned off, so that means autocorrect should be enabled. |
| 16:51.24 | p3nguin | But how does it work? |
| 16:51.50 | p3nguin | If I misspell a word, it doesn't automatically change it when I hit space and it doesn't fix it when I actually send it to the channel. |
| 16:51.54 | p3nguin | Seems like it doesn't work. |
| 16:52.28 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 16:52.31 | *** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 16:52.41 | a1fa | wow -- soo frustrated with broadvoice support |
| 16:53.02 | a1fa | find first 10 people that walk in the door, and give them access to manage peoples account |
| 16:55.04 | a1fa | ~us-p |
| 16:55.16 | a1fa | what was that command to get a list of reputable providers |
| 16:55.20 | a1fa | may need to take bv out of it |
| 16:57.52 | *** join/#asterisk leifmadsen (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:57.52 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:58.23 | *** join/#asterisk ruied (~AndChat66@33.100.103.87.rev.vodafone.pt) |
| 17:00.27 | Qwell | ~itsplist-us |
| 17:00.27 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 17:01.20 | a1fa | Qwell: how to remove a provider from the list? |
| 17:02.08 | p3nguin | haha |
| 17:02.21 | a1fa | whats funny? |
| 17:02.36 | p3nguin | If enough people complain about bv, I'll remove it myself. |
| 17:02.52 | a1fa | have they not? |
| 17:03.10 | p3nguin | I don't remember anyone else complaining. |
| 17:03.12 | [TK]D-Fender | a1fa, It's just you. |
| 17:03.18 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 17:03.28 | a1fa | i've been with this people for 8 years |
| 17:03.37 | a1fa | and they foobar my account |
| 17:03.50 | a1fa | now i cant get it back |
| 17:04.05 | a1fa | i guess its just me |
| 17:06.28 | *** join/#asterisk ruied (~AndChat66@114.32.166.178.rev.vodafone.pt) |
| 17:06.58 | a1fa | Teliax unlimited inbound looks like the way to go |
| 17:07.50 | [TK]D-Fender | a1fa, FUBAR'd how? Can't get back why? |
| 17:07.52 | a1fa | is number porting any better/faster than it used to be? |
| 17:08.46 | a1fa | [TK]D-Fender : i had it on auto-pay, and my card got compromised, got a new card, never updated the one on broadvoice, broadvoice send a few notes that never got to me.. |
| 17:09.17 | [TK]D-Fender | a1fa, So basically you didn't pay your bills, didn't get the notices, and they dumped your ass. |
| 17:09.20 | a1fa | technically my fault |
| 17:09.25 | a1fa | yes |
| 17:10.21 | [TK]D-Fender | a1fa, http://tinyurl.com/dxf8ozf |
| 17:10.46 | *** join/#asterisk TechSmurf (~jdaniel@unaffiliated/techsmurf) |
| 17:10.47 | a1fa | +1 |
| 17:11.21 | a1fa | how easy is it to move providers and take your number with you? |
| 17:11.40 | Kobaz | it's as easy as getting out your wallet |
| 17:11.51 | a1fa | $$$? |
| 17:11.54 | a1fa | how much |
| 17:12.01 | Kobaz | usually like 10-20 |
| 17:12.08 | Kobaz | per number |
| 17:12.13 | Kobaz | unless you do a mass-port |
| 17:12.16 | Kobaz | then you can work out a deal |
| 17:12.18 | Qwell | and 6 weeks, while the old provider dicks you around. heh |
| 17:12.27 | a1fa | not bad -- wow.. 6 weeks? |
| 17:12.29 | Kobaz | it shouldn't be more than like 10 days |
| 17:12.30 | a1fa | still as bad as it was |
| 17:12.31 | Qwell | sometimes :p |
| 17:12.38 | Kobaz | if the provider is not an ass |
| 17:12.41 | a1fa | i remember... 7 years ago |
| 17:12.48 | a1fa | it was impossible to port a number |
| 17:12.50 | Qwell | I've seen horror stories. |
| 17:13.08 | a1fa | there should be a did portal |
| 17:13.14 | a1fa | you buy a number at birth |
| 17:13.22 | a1fa | and move it to a different provider yourself |
| 17:13.30 | a1fa | a1fa for president anyone? |
| 17:13.46 | a1fa | we'll use SSN for dialing ;) |
| 17:13.59 | a1fa | talk about identitity management |
| 17:14.10 | Qwell | While we're at it, let's merge phone #s and license #s |
| 17:14.19 | a1fa | agreed :) |
| 17:14.22 | Qwell | I want to yell at the jerk in front of me when he cuts me off. |
| 17:14.31 | a1fa | SS > DL# > P# |
| 17:17.02 | Nivex | "I am not a number. I am a free man!" |
| 17:17.18 | a1fa | phonepower? |
| 17:17.28 | a1fa | they want to move my account to phonepwer |
| 17:23.12 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 17:23.38 | *** join/#asterisk ruied (~AndChat66@114.32.166.178.rev.vodafone.pt) |
| 17:23.57 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 17:24.12 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 17:24.39 | SeRi | p3nguin: call my itad and let it ring for like 10 seconds please. Doing a test. |
| 17:26.12 | a1fa | anybody know anything about this phonepower ? |
| 17:26.19 | a1fa | ~phonepower |
| 17:26.32 | [TK]D-Fender | a1fa, Never heard of |
| 17:26.48 | a1fa | phonepower owns broadvoice |
| 17:27.01 | a1fa | they moved my account from broadvoice over to phonepower |
| 17:27.10 | a1fa | because it got fubared |
| 17:28.33 | SeRi | a1fa: http://www.dslreports.com/forum/remark,27143193?hilite=phonepower |
| 17:28.40 | *** join/#asterisk navaismo (~navaismo@189.144.120.135) |
| 17:29.00 | SeRi | and |
| 17:29.04 | SeRi | http://www.dslreports.com/forum/remark,22887509?hilite=phonepower |
| 17:29.08 | SeRi | and it keeps going |
| 17:29.22 | SeRi | they dont seem to be a popular itsp on dslr |
| 17:29.45 | SeRi | But that does not mean that they are not reputable |
| 17:29.50 | *** part/#asterisk Aethrs (~secret@static-71-251-98-122.tampfl.fios.verizon.net) |
| 17:29.53 | SeRi | do some resarch |
| 17:30.33 | a1fa | well they just moved my account over there |
| 17:31.17 | a1fa | BYOD; same plan and pricing as broadvoice on phonepower |
| 17:31.23 | a1fa | sounds like i'll be making a call to teliax |
| 17:31.27 | a1fa | to see if i can rescue my digits |
| 17:34.35 | SeRi | good luck |
| 17:35.09 | SeRi | ha. I just called one the comcast managers and they for got they had to come out here. what a joke. |
| 17:38.15 | a1fa | i'm migrating over to teliax |
| 17:38.19 | a1fa | pay as you go |
| 17:38.28 | a1fa | just need to make sure i setup good inbound filters ;) |
| 17:38.35 | a1fa | you need to be on the whitelist for inbound call |
| 17:38.36 | jpsharp | SeRi: And you're surprised? |
| 17:38.52 | SeRi | jpsharp: the person I been looking for. |
| 17:38.53 | a1fa | i went from local cable company to u-verse |
| 17:38.57 | SeRi | I move to comcast business jpsharp |
| 17:38.58 | a1fa | speeds nearly trippled |
| 17:39.10 | a1fa | i was getting 1Mbps during prime time on local cable DOCSIS 3.0 |
| 17:39.20 | a1fa | versus 7Mpbs on UVERSE |
| 17:39.24 | SeRi | a1fa: Thats a none issue for me. |
| 17:39.29 | a1fa | but I am paying for 20 |
| 17:39.37 | a1fa | these internetproviders are a sham |
| 17:39.45 | a1fa | even on our business OC192 they lie |
| 17:39.47 | SeRi | jpsharp: so now I am in comcast business and my line is even worst |
| 17:40.01 | a1fa | you only get 80% of your OC192 due to overhead :) |
| 17:40.01 | SeRi | a manager is suppose to come by and drop a RG11 line |
| 17:40.13 | *** join/#asterisk NewUser10 (32522942@gateway/web/freenode/ip.50.82.41.66) |
| 17:40.26 | SeRi | another tech supose to come by with a new modem |
| 17:40.45 | SeRi | jpsharp: did you had issues with sip on comcast biz? |
| 17:41.59 | NewUser10 | Hi, How does Asterisk handle a in-progress phone call if the VOIP phone I'm talking on looses it's connection to the Asterisk server? |
| 17:42.38 | WIMPy | NewUser10: See rtptimeout and session timers. |
| 17:44.04 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 17:47.40 | *** join/#asterisk blitzrage (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:47.41 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 17:51.46 | *** join/#asterisk MZXGiant (mzxgiant@cpe-24-93-21-126.rochester.res.rr.com) |
| 17:52.30 | MZXGiant | Good afternoon, folks. I was hoping someone could help an asterisk n00b like myself. I'm having a problem on my asterisk installation on Ubuntu 11.04 with app_stack not loading on start; I have to go into the CLI and manually load it |
| 17:52.46 | *** join/#asterisk Sicelo (Sicelo@unaffiliated/sicelo) |
| 17:52.47 | MZXGiant | Is there a place where I can fix this? (I tried just adding preload => app_stack.so to the config with no luck) |
| 17:52.52 | MZXGiant | *to the modules.config |
| 17:52.58 | MZXGiant | ... *modules.conf |
| 17:53.10 | MZXGiant | Need... more... coffee... |
| 17:53.29 | NewUser10 | Is there a way to make Asterisk put a call on hold with a specific message when the RTP timesout, instead of the automatic hangup? |
| 17:55.08 | a1fa | it will never hangup then |
| 17:55.18 | a1fa | thats one of the ways a phone is hangup |
| 17:57.09 | NewUser10 | Our internet is a bit flakey and we experience dropped calls at least once a week with our current VOIP provider. It usually comes back within a few minutes, but still anoying. |
| 17:57.50 | MZXGiant | Changed "preload" to "load" -- still no effect =/ |
| 17:57.57 | NewUser10 | I was hoping Asterisk could gracefully handle this (put call on hold when phone drops). Maybe another timer (RTPTimeoutHangup?) could be used to auto hangup those calls... |
| 17:58.09 | MZXGiant | If I don't get this loaded by default, I can't make any calls, even ext<->ext |
| 17:58.16 | MZXGiant | Hrmrm. :P |
| 17:58.43 | *** join/#asterisk ruied (~AndChat66@114.32.166.178.rev.vodafone.pt) |
| 17:59.23 | NewUser10 | Or even better... when the VOIP phone drops, put the call on hold, call our cell phone, and resume the call there, all automatically... |
| 17:59.35 | NewUser10 | We usually just finish up on the cell phone anyways... |
| 17:59.41 | NewUser10 | Would be nice for automation :) |
| 17:59.45 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 18:00.03 | NewUser10 | Would such a behaviour be hard to program into Asterisk? |
| 18:00.28 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 18:01.21 | MZXGiant | "undefined symbol: ast_agi_unregister" looks like the culprit for why it's not auto-loading |
| 18:03.04 | Qwell | MZXGiant: What version of Asterisk? |
| 18:04.41 | MZXGiant | Qwell; Connected to Asterisk 1.8.4.4~dfsg-2ubuntu1.1 |
| 18:05.01 | MZXGiant | It's the version from the oneric aptitude repo |
| 18:05.02 | Qwell | why so old? |
| 18:05.10 | Qwell | and, do you have res_agi loaded? |
| 18:05.35 | MZXGiant | It's not referenced in modules.conf |
| 18:05.57 | MZXGiant | ffff |
| 18:06.09 | MZXGiant | Sorry, didn't even realize the aptitude tree on this system is this out of date |
| 18:06.21 | MZXGiant | looks like one of the junior sysops is getting a talking to this week =/ |
| 18:09.25 | MZXGiant | Looks like newest available in apt is... 1.8.11.1, Qwell |
| 18:09.27 | MZXGiant | that better? :) |
| 18:09.32 | Qwell | Nope. |
| 18:09.35 | SeRi | still old |
| 18:10.17 | MZXGiant | D'oh. |
| 18:11.00 | MZXGiant | I was really hoping the universe dpkgs would be up to date =/ |
| 18:11.02 | MZXGiant | oh well |
| 18:12.21 | MZXGiant | Just FYI (I'm following the instruction manual now), I'm getting this: |
| 18:12.23 | MZXGiant | root@riker:~# add-apt-repository "deb-src http://packages.asterisk.org/deb `lsb_release -cs` main" |
| 18:12.23 | MZXGiant | Error: 'deb-src http://packages.asterisk.org/deb oneiric main' invalid |
| 18:12.35 | MZXGiant | (command came from https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-APT(Debian%2FUbuntu) ) |
| 18:12.53 | Sicelo | such is the pain of debian-based stuff. then again, you can compile the latest on your own |
| 18:12.58 | MZXGiant | but the binary repo works |
| 18:12.58 | *** join/#asterisk jpcansa (~JP@200.91.100.35) |
| 18:13.45 | *** join/#asterisk fabsoft (5ff2618d@gateway/web/freenode/ip.95.242.97.141) |
| 18:13.47 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 18:14.33 | fabsoft | hi all |
| 18:14.36 | SeRi | MZXGiant: looks like a senior sysops is getting a talk this week. |
| 18:14.40 | SeRi | ;) |
| 18:14.42 | fabsoft | is there a way to use "+" in regex expression ? |
| 18:14.56 | fabsoft | i would to strip + from callerid |
| 18:15.07 | MZXGiant | SeRi; Hahaha |
| 18:15.12 | fabsoft | normal \ escape doesn't work |
| 18:15.13 | Sicelo | lol SeRi |
| 18:15.18 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 18:15.47 | MZXGiant | I leave most of the dev(/low impact) systems to the junior staff and just trust they're trying things like "update the apt tree" |
| 18:16.06 | MZXGiant | so when things get pushed to me for production implementation, I take for granted that everyone below me has done their job |
| 18:16.11 | MZXGiant | guess that's not a good idea ;) |
| 18:16.22 | SeRi | never has |
| 18:16.37 | SeRi | all ways double check the work of others |
| 18:16.41 | SeRi | even your senior peerrs |
| 18:16.46 | SeRi | :) |
| 18:16.54 | SeRi | s/peerrs/peers/ |
| 18:17.00 | SeRi | meh |
| 18:17.01 | MZXGiant | I check my boss' work but that's because he's a MBA weenie |
| 18:17.03 | MZXGiant | :P |
| 18:17.12 | SeRi | lol |
| 18:17.31 | MZXGiant | (no offense to people with MBAs-- I respect the program, just not people who think an MBA is sufficient to be a "hands-on CTO") |
| 18:18.00 | *** join/#asterisk fneuwald (~fneuwald@201.86.128.17.static.gvt.net.br) |
| 18:18.23 | [TK]D-Fender | fabsoft, No need for Regex for that |
| 18:19.19 | MZXGiant | Looks like just upgrading to the node in the top of the apt tree fixed the problem |
| 18:19.21 | MZXGiant | sighs |
| 18:19.23 | fneuwald | Hi folks. I'm developing a outbound TTS application, with AMI originate. The TTS start even when the destination don't answer the call. There is some way to implement "wait answer", or something like? |
| 18:19.26 | MZXGiant | Thanks guys :P |
| 18:21.27 | SeRi | cya MZXGiant |
| 18:25.08 | fneuwald | I tried WaitForNoise on dialplan, but got no success. |
| 18:29.41 | *** join/#asterisk Rico29 (~rico@oceanet-telecom-fttb-129-2.olm.fr) |
| 18:29.48 | fabsoft | [TK]D-Fender: what do you suggest ? |
| 18:29.51 | *** join/#asterisk imox (~imox@91-66-32-57-dynip.superkabel.de) |
| 18:30.05 | [TK]D-Fender | fabsoft, Where does the + occur? |
| 18:30.18 | fabsoft | maybe, in the callerid(num) |
| 18:30.35 | fabsoft | i have to check if it exists.. |
| 18:30.53 | fabsoft | this way: exten => s,n,ExecIf($[${REGEX("^\+" ${CALLERID(NUM)})}]?Set(CALLERID(NUM)=${CALLERID(NUM):1})) |
| 18:31.24 | [TK]D-Fender | fabsoft, easy answer : core show function CUT |
| 18:32.02 | fabsoft | [TK]D-Fender: thank you i'll see soon |
| 18:34.35 | *** join/#asterisk tuxfarva (hidden-use@2620:3f:8000::51) |
| 18:36.02 | fneuwald | I'm developing a outbound TTS application, with AMI originate. The TTS start even when the destination don't answer the call. There is some way to implement "wait answer", or something like? |
| 18:36.27 | fneuwald | the tts server connects via sip to the asterisk server, and the asterisk server has a connection to pstn using mfcr2 |
| 18:36.29 | Qwell | fneuwald: Yes, we saw your question the first time. |
| 18:36.36 | fneuwald | good. |
| 18:36.39 | fneuwald | tks |
| 18:39.09 | fneuwald | Qwell: do you know any solution for this? |
| 18:39.16 | *** part/#asterisk fneuwald (~fneuwald@201.86.128.17.static.gvt.net.br) |
| 18:39.23 | *** join/#asterisk fneuwald (~fneuwald@201.86.128.17.static.gvt.net.br) |
| 18:39.42 | Qwell | fneuwald: If I did, I would have answered you. |
| 18:39.42 | fneuwald | sorry. command + w on wrong window :-| |
| 18:40.32 | [TK]D-Fender | fneuwald, The only reason your dialplan should execute is if it did answer. Which it apparently did |
| 18:41.26 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 18:42.41 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:42.41 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:44.21 | fabsoft | [TK]D-Fender: exten => s,n,Set(CALLERID(NUM)=${CUT(,CALLERID(NUM),+,1)}) this works for you ? |
| 18:45.14 | fneuwald | [TK]D-Fender: take a look at http://pastebin.com/mSGBh8S6 - you'll see that the play starts even before answer. |
| 18:45.16 | [TK]D-Fender | fabsoft, I think you should look at your parms for CUT again very closely... |
| 18:45.38 | fabsoft | [TK]D-Fender: heeh the comma |
| 18:45.54 | [TK]D-Fender | fneuwald, Channel Local/999@tts-82f8;1 was answered. |
| 18:45.55 | [TK]D-Fender | ^ |
| 18:46.15 | [TK]D-Fender | fneuwald, In order for it to wait for noise ... it has to have ANSWERED |
| 18:46.26 | [TK]D-Fender | fneuwald, You are not thinking this process through properly |
| 18:46.33 | [TK]D-Fender | fabsoft, Indeed. |
| 18:46.45 | fneuwald | so, any other workaround for this? |
| 18:47.58 | [TK]D-Fender | fneuwald, Your implementation is BACKWARDS. |
| 18:48.15 | [TK]D-Fender | You need to call your SIP end in the Channel, and THEN dump them into that other context |
| 18:49.26 | fneuwald | i'll try to change here. just a moment |
| 18:52.56 | *** join/#asterisk Sicelo- (~user@unaffiliated/sicelo) |
| 18:53.53 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:54.29 | fabsoft | [TK]D-Fender: it does not worl |
| 18:54.51 | [TK]D-Fender | fabsoft, And you are nos showing us your new code and your actual attempt showing the before & after... |
| 18:54.53 | [TK]D-Fender | ~pb |
| 18:54.53 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 18:54.56 | [TK]D-Fender | ^^ your friend |
| 18:55.51 | *** join/#asterisk Sicelo- (~user@unaffiliated/sicelo) |
| 18:56.04 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 18:56.07 | SeRi | ok first tech is here |
| 18:56.10 | SeRi | disconnecting |
| 18:56.13 | fabsoft | exten => s,n,Set(CALLERID(NUM)=${CUT(,CALLERID(NUM),+,1)}) when + are matched it cuts everythink |
| 18:57.53 | fneuwald | [TK]D-Fender: thanks man. solved. :-) |
| 18:58.03 | fneuwald | [TK]D-Fender: where should I put the $ for donation? :-) |
| 18:58.05 | fabsoft | [TK]D-Fender: REPLACE function also should works, but --> No application 'REPLACE' for extension |
| 18:58.43 | Qwell | REPLACE isn't an application. Are you using it properly? |
| 18:58.48 | [TK]D-Fender | fneuwald, I do use Paypal if you feel so inclined |
| 18:58.59 | [TK]D-Fender | Qwell, evidently not. |
| 18:59.54 | [TK]D-Fender | <fabsoft> exten => s,n,Set(CALLERID(NUM)=${CUT(,CALLERID(NUM),+,1)}) when + are matched it cuts everythink <- still an extra "," |
| 19:00.59 | fneuwald | [TK]D-Fender: thanks. i'll send you info about paypal on pvt. |
| 19:01.16 | Qwell | fabsoft: You're also asking for everything before the +. |
| 19:01.51 | fabsoft | [TK]D-Fender: Qwell i've solved with: exten => s,n,Set(CALLERID(NUM)=${REPLACE(CALLERID(NUM),+,)}) |
| 19:01.56 | fabsoft | thank you all |
| 19:04.29 | *** part/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 19:07.57 | fabsoft | ok.. anyone has used h323 recently ?? |
| 19:09.01 | *** join/#asterisk italorossi (~italoross@189.124.196.68) |
| 19:14.09 | qakhan | hi all |
| 19:14.22 | qakhan | how to setup annousment in queue |
| 19:14.37 | qakhan | how to setup annoucment in queue* |
| 19:18.57 | *** join/#asterisk mobile_gordita (~Robert@66-87-95-14.pools.spcsdns.net) |
| 19:19.10 | *** join/#asterisk F2Knight (~TuxPowere@70-89-188-5-or.portland.hfc.comcastbusiness.net) |
| 19:22.39 | F2Knight | Q: about dahdi configuration. x2 Digium cards. 1 Quad, 1 Dual, dahdi_genconf for the setup, and the Quad works, but the dual is in red alarm. Moved the cables on between the to cards and the quad does not go in to red alarm, which leads me to believe it is a config error in dahdi_genconf default, I tried setting the timing source on the dual card to span=5,1,0,esf,b8zs. Any suggestions ? rebooted it is still in red al |
| 19:28.25 | *** join/#asterisk DelphiWorld (~VoCloud@openvpn/user/DelphiWorld) |
| 19:28.30 | *** part/#asterisk DelphiWorld (~VoCloud@openvpn/user/DelphiWorld) |
| 19:30.36 | *** join/#asterisk AviMarcus (~avi@bzq-79-183-236-173.red.bezeqint.net) |
| 19:30.40 | *** part/#asterisk AviMarcus (~avi@bzq-79-183-236-173.red.bezeqint.net) |
| 19:32.17 | qakhan | how to setup annoucment in queue |
| 19:33.19 | *** join/#asterisk Sicelo900 (~user@unaffiliated/sicelo) |
| 19:34.49 | a1fa | ah |
| 19:34.54 | a1fa | i initiated a port of my numbeer |
| 19:34.59 | a1fa | to teliax |
| 19:35.50 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 19:36.30 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 19:36.40 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 19:37.52 | navaismo | qakhan, http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf |
| 19:38.41 | NewUser10 | a1fa: Have you checked out FlowRoute? They have better rates for pay as you go... |
| 19:39.07 | a1fa | i looked into them recently |
| 19:39.18 | SeRi | I use flowroute. They are ok... |
| 19:39.27 | SeRi | I use voip.ms as my main itsp |
| 19:39.35 | SeRi | flowroute is backup |
| 19:39.55 | SeRi | p3nguin: so... no dice still. |
| 19:40.13 | SeRi | p3nguin: Now they say that the issue is somewhere in the naighborhood. |
| 19:40.28 | SeRi | My upload still crap. |
| 19:40.38 | SeRi | and I mean crap as in caca |
| 19:41.10 | *** join/#asterisk spditner (~simon@206-248-134-234.dsl.teksavvy.com) |
| 19:42.26 | qakhan | navaismo i setup as same as it mentioned in this link |
| 19:42.31 | qakhan | but its not working |
| 19:42.36 | a1fa | i'm going to pay as you go + 2 dids :) |
| 19:42.58 | SeRi | qakhan: I hired navaismo. he can setup a call center for you :) |
| 19:44.50 | pabelanger | ANybody attempted zoiper on ubuntu 12.04? |
| 19:46.02 | [TK]D-Fender | qakhan, Show us the problem along with your configs & file dumps to prove that things are where you claim they are. |
| 19:46.27 | [TK]D-Fender | qakhan, Saying "It doesn't work" does not offer anything for us to help you with. Don't just say "it doesn't work" ... SHOW US |
| 19:46.45 | a1fa | wow.. DID prices went up.. it used to be $1.99/DID.. now $4.99 :( |
| 19:47.03 | NewUser10 | How is it that Canada starts at $0.0052, but the USA 48 starts at $0.105 (for voip.ms) ? |
| 19:47.17 | *** join/#asterisk leifmadsen (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:47.17 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:47.26 | *** join/#asterisk vince_ (~quassel@2620:8b:0:f065:a8e0:c162:983f:eac0) |
| 19:47.58 | [TK]D-Fender | NewUser10, link please... |
| 19:48.18 | NewUser10 | [TK]D-Fender: http://www.voip.ms/ |
| 19:49.09 | Venaran | canadian dids are still 0.99 or 1.99 |
| 19:49.12 | Venaran | which ones are 4.99? |
| 19:50.38 | a1fa | US DID |
| 19:51.04 | [TK]D-Fender | NewUser10, Yup... that is somewhat insane.... |
| 19:51.18 | Venaran | i just picked miami florida as a random spot, and DIDs there are 0.99 |
| 19:51.39 | Venaran | the 4.95 plan are the flat rate plans |
| 19:52.42 | *** join/#asterisk Sicelo900 (~user@unaffiliated/sicelo) |
| 19:53.44 | a1fa | Venaran : teliax? |
| 19:54.17 | qakhan | [TK]D-Fender here is my dial plan and queues.conf |
| 19:54.17 | qakhan | http://pastebin.com/nUuaZ5Ds |
| 19:54.18 | Venaran | Im sorry a1fa I thought you were talking about voip.ms |
| 19:54.39 | *** join/#asterisk bchia (~Adium@nat/digium/x-qrjablpzdxbgbhfk) |
| 19:58.46 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 20:00.07 | [TK]D-Fender | qakhan, And what announcement are you expecting to happe? Why don't we see your CALL attempt in there as well? |
| 20:00.57 | p3nguin | seri: That's pretty shitty. I take it you don't need that test call anymore. |
| 20:01.13 | qakhan | my call goes to queue and annoucement does play if i put call on hold for 5 or 10 mins |
| 20:07.24 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 20:07.58 | navaismo | qakhan, did you tried to set all announce setting within your queue definition and no outside? |
| 20:08.23 | qakhan | yes |
| 20:08.42 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 20:08.54 | qakhan | its even does not show any message in cli of annoucement |
| 20:09.19 | navaismo | and reloaded? as [TK]D-Fender said, show you actual configs and the cli output and more important the announcement exist? I mean the recording? |
| 20:09.50 | SeRi | p3nguin: The test call is for a cdr change I mad and wanted to see if it is working |
| 20:09.57 | SeRi | p3nguin: so yes if you can :) |
| 20:10.11 | [TK]D-Fender | qakhan, WHAT announcement? I do not see you having configured a specific file to play. You haven't explains WHAT you are expecting to hear and you aren't showing us the call. |
| 20:10.17 | qakhan | does asterisk not has defualt annoucement recording? i think it has |
| 20:10.36 | qakhan | wait plz |
| 20:11.08 | navaismo | ¬¬ |
| 20:11.27 | [TK]D-Fender | qakhan, assume != think |
| 20:11.56 | [TK]D-Fender | qakhan, And you are not showing a complete picture. While you're at it, clean up your config by removing all the commented out junk. |
| 20:12.41 | SeRi | looks like I might be switching to flow route |
| 20:12.56 | SeRi | flowroute is cheaper than voip.ms right now |
| 20:14.28 | qakhan | here is my cli |
| 20:14.34 | qakhan | http://pastebin.com/FBQYN8i1 |
| 20:16.04 | [TK]D-Fender | qakhan, You didn't set the frequency |
| 20:16.33 | [TK]D-Fender | or does it apply... hrm |
| 20:16.42 | [TK]D-Fender | qakhan, Clean up your config and repost |
| 20:17.11 | NewUser10 | Can anyone recommend a FXS ATA ? I am looking at the HT701 |
| 20:17.30 | navaismo | and validate if the sound file exist |
| 20:17.37 | [TK]D-Fender | NewUser10, What are yo planning on doing with it? |
| 20:17.40 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 20:18.33 | qakhan | [TK]D-Fender here is clean post |
| 20:18.34 | qakhan | http://pastebin.com/rdi0ddZ5 |
| 20:19.17 | [TK]D-Fender | qakhan, ;announce-holdtime = yes <- commented out. It will not announce position/holdtime |
| 20:20.28 | [TK]D-Fender | qakhan, ;periodic-announce-frequency = 15 <- commented out. It will not announce the "periodic stuff either |
| 20:20.59 | qakhan | do i comment it? or uncomment? |
| 20:20.59 | *** join/#asterisk funky1 (~jarvis@ip51cf100e.direct-adsl.nl) |
| 20:24.50 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 20:25.03 | Snivets | what's the sip-notify command for rebooting snom m9s, anyone know? i tried snom-reboot and just straight up reboot and Asterisk says it failed. just failed, no real specifics. |
| 20:25.43 | SeRi | p3nguin: a network eng just call me from comcast |
| 20:25.55 | SeRi | "they are working on the issue" |
| 20:26.12 | SeRi | At least this is having more attention than when it was residential for sure |
| 20:27.23 | *** join/#asterisk leifmadsen (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:27.23 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:29.18 | qakhan | [TK]D-Fender i uncommented ;announce-holdtime = yes and ;periodic-announce-frequency = 15 |
| 20:29.23 | qakhan | but still not working |
| 20:29.59 | navaismo | reloaded and again the file exist? |
| 20:30.11 | F2Knight | Does anyone know if your running multiple t1 cards, if each card needs its own timing source? |
| 20:30.31 | WIMPy | F2Knight: Unless you use a timing cable. |
| 20:31.22 | funky1 | hi all :) have some audio problems with asterisk freepbx, probably nat issue, incoming calls audio works fine both ways, when doing outgoing call the outgoing audio drops after a minute or so and comes back after 15 secs goes again, comes back... i have rtp ports forwarded to my asterisk and configured asterisk accordingly but my port 5060 is closed, asterisk and the extension from which i make the outgoing call are on same lan behind router, |
| 20:31.23 | funky1 | do i need to open my inbound 5060 port or any other ideas what the problem might be? |
| 20:31.37 | qakhan | navaismo i reloaded |
| 20:32.10 | qakhan | where i put file name |
| 20:34.55 | NewUser10 | [TK]D-Fender: make my analog phone an extension of the Asterisk system |
| 20:34.58 | F2Knight | WIMPy: Not sure what that is so I can be sure I am not using it :-) Just 2 digium T1 cards in a single machine. |
| 20:35.05 | navaismo | funky1, try #freepbx and folllow the nat settings instructions |
| 20:35.31 | WIMPy | F2Knight: Then you need exactely one timing source per card. |
| 20:35.35 | NewUser10 | [TK]D-Fender: just looking for something simple, nothing fancy needed |
| 20:35.46 | [TK]D-Fender | NewUser10, For a regular phone : Linksys PAP2T-NA |
| 20:35.51 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 20:36.16 | funky1 | navaismo: i got freepbx installed, but i find the nat settings not clear, not sure what i need to set in there, yes, no never or route, any tips? |
| 20:38.15 | navaismo | if you are behind nat usually is yes |
| 20:39.37 | NewUser10 | [TK]D-Fender: Thanks! |
| 20:40.15 | F2Knight | WIMPy: http://pastebin.com/2trwxSLU |
| 20:40.34 | funky1 | navaismo: i have actually tried all available settings there with all of them, asterisk behaves in same way, inbound calls=audio both ways, outbound calls=inbound audio fine, outbound audio drops, comes back, drops... |
| 20:40.42 | qakhan | [TK]D-Fender any update |
| 20:40.44 | funky1 | other suggestions? |
| 20:40.49 | F2Knight | Span 5 is the issue it is the first span on the second card, I tried changing the timing, but still red alrams |
| 20:41.16 | qakhan | or i need to put any option in Queue application in dial plan |
| 20:41.52 | WIMPy | F2Knight: red is no signal. Fo far away from any timing issues. |
| 20:42.05 | vince_ | if you where on asterisk 1.6.2 and had to upgrade would you go to 1.8 or to 11? |
| 20:42.33 | Venaran | do you have time to test it out first? |
| 20:44.08 | *** join/#asterisk fritz09 (~Adium@pop1-224.catv.wtnet.de) |
| 20:44.16 | vince_ | have another server that is basically a copy of live but not connected to any dahdi so can't test further then making sure everything runs |
| 20:44.41 | [TK]D-Fender | qakhan, I'm not seeing updated configs, no proof of files being where they should, no new call debug to match. I am tired of having to ask for these each and every time you make changes while trying to fix a problem. I do not have time to waste on this. |
| 20:46.39 | qakhan | i sent you every thing |
| 20:49.57 | [TK]D-Fender | qakhan, You didn't. You made changes and did NOT show them. I NEVER saw and file lists to prove any files you "think" should be playing even have a chance to succeed. I am not wasting my time on this. |
| 20:50.05 | [TK]D-Fender | moves on to more productive matters |
| 20:50.53 | F2Knight | WIMPy: except when I plug that line in to span 1 − 4 it does not red alarm |
| 20:50.58 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:51.20 | F2Knight | thus the confusion |
| 20:51.24 | Venaran | well vince I personally am still on 1.8, so I would suggest that, but I really couldnt give you a good set of reasons why I am still there, other than because upgrading is a huge hassle |
| 20:53.39 | Snivets | has serious trouble figuring out how assigning snoms to base stations actually works |
| 20:55.27 | vince_ | what is the best way to upgrade asterisk? just ./configure make make install and hope for the best? there really are not any guides out there for upgrading |
| 20:56.10 | Venaran | do it on your other machine firest |
| 20:56.15 | Venaran | make sure your config file can carry over |
| 20:56.16 | Sicelo900 | i'd backup my .confs first, lol |
| 20:56.42 | Venaran | you dont want to change to a new version and find you have deprecated commands |
| 20:58.49 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 21:04.57 | *** join/#asterisk MZXGiant (mzxgiant@cpe-24-93-21-126.rochester.res.rr.com) |
| 21:05.11 | MZXGiant | Good afternoon, all -- I'm back, with another question which will likely be as stupid as the first :) |
| 21:05.47 | MZXGiant | I got a SIP trunk set up for incoming/outgoing calls ... I call into the DID, I see Asterisk pick up in the CLI, and then it redirects to the directory, and I see this: |
| 21:05.52 | MZXGiant | <PROTECTED> |
| 21:05.52 | MZXGiant | [Nov 29 16:02:28] WARNING[19962]: file.c:663 ast_openstream_full: File cdir-welcome does not exist in any format |
| 21:05.52 | MZXGiant | [Nov 29 16:02:28] WARNING[19962]: file.c:958 ast_streamfile: Unable to open cdir-welcome (format 0x4 (ulaw)): No such file or directory |
| 21:05.52 | MZXGiant | <PROTECTED> |
| 21:06.22 | MZXGiant | If I go find the recordings from my GUI (I'm using FreePBX), I can select "cdir-welcome" and listen to it |
| 21:12.32 | Qwell | Asterisk doesn't plan files from the recordings directory. |
| 21:12.34 | Qwell | play* |
| 21:13.01 | MZXGiant | Hm! I wonder how this is supposed to work, then. :P |
| 21:13.09 | MZXGiant | googles for a manual |
| 21:13.34 | Qwell | ~book |
| 21:13.35 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 21:14.51 | MZXGiant | Thanks :) |
| 21:15.00 | MZXGiant | I'm kind of learning-by-breaking-it for this |
| 21:15.14 | MZXGiant | I chose Wireless Antenna Theory over IP Telefony for my concentration in college :P |
| 21:17.37 | Nivex | ARRL Antenna Handbook ftw |
| 21:18.37 | jpsharp | Majoring in antenna development? |
| 21:19.39 | qakhan | [TK]D-Fender thanks for all your help, as you always help me :) |
| 21:19.45 | MZXGiant | Nivex ^5 |
| 21:20.26 | MZXGiant | jpsharp; Just had an option and the idea of sitting for 11 hours in a telefony lab every week trying to make the antiquated Meridian cabinets work didn't interest me |
| 21:20.40 | MZXGiant | So instead I built a 40-element Yagi for my design project :P |
| 21:20.58 | qakhan | issue resolved after put announce-frequency = 10 periodic-announce-frequency = 5 announce-holdtime = yes queue-holdtime = queue-holdtime |
| 21:21.05 | qakhan | in queue |
| 21:22.04 | jpsharp | MZXGiant: Oh, can't say I blame you. I've fondled enough option 11 systems to dislike them. And what frequency Yagi? |
| 21:22.51 | *** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 21:23.31 | MZXGiant | jpsharp; 2.4GHz |
| 21:24.01 | MZXGiant | We had a range test at the end of the year using some REALLY old transmission equipment that had an SMA connector to the antenna |
| 21:24.14 | MZXGiant | and we essentially shot clean 2.4GHz signal -- 802.11 base standard |
| 21:24.55 | MZXGiant | the measurements for that took forever, and then we realized we had massive interference from our center-beam which we opted not to kiln-dry to save time |
| 21:25.33 | jpsharp | Pesky 2.4Ghz not liking water. |
| 21:25.35 | MZXGiant | But I ended up making ceramic insulators for all the copper rods to keep them from passively grounding inside the wooden center-beam |
| 21:26.04 | MZXGiant | Got a lot of use from my calipers those weeks |
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| 21:27.17 | jpsharp | I believe it. I pondered building a 1691Mhz Yagi to receive GOES weather data rather than using a dish, but I don't think I could build it within tolerances. |
| 21:27.42 | MZXGiant | It's a PITA |
| 21:28.02 | jpsharp | So I'll stick with trying to find a 1.2M dish. |
| 21:28.05 | MZXGiant | That dipole at the base alone is a huge pain to get the gap right between your radiating element |
| 21:28.33 | MZXGiant | Let alone angular consistency and the reflector distances and lengths |
| 21:28.43 | MZXGiant | I wouldn't do that again. I'm proud of it, but I wouldn't do it again :P |
| 21:28.54 | jpsharp | Yep. Had hard enough time building a 440Mhz yagi. |
| 21:29.44 | MZXGiant | Heh, I still have a picture of myself testing this thing on my g+ |
| 21:30.12 | MZXGiant | PMed link, jp -- prefer not to spam the channel with it :) |
| 21:32.10 | jpsharp | You can barely see the elements. Heh. |
| 21:32.33 | MZXGiant | This is pre-ceramic insulators |
| 21:34.42 | *** join/#asterisk danfromuk (IceChat77@unaffiliated/danfromuk) |
| 21:44.08 | *** join/#asterisk paryl (~par@168.215.170.202) |
| 21:46.33 | paryl | hi everyone, i'm having trouble with meetme conferences. i'm using realtime, and an external app creates the conference in the database. users dial in using the chosen passcodes. if they all use the admin code, everything is ok, but if the users use the regular user passcode, no one can hear them. i'm not setting any options in the dialplan, it's just <exten>,1,MeetMe |
| 21:47.16 | paryl | is there something i'm missing? i don't know what else to check |
| 21:50.54 | SeRi | this is depressing. |
| 21:52.57 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
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| 22:06.56 | myyrdin | paryl: you don't have enter muted set do you? try having them press * and unmute themselves |
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| 22:56.05 | cusco | hi |
| 22:56.15 | cusco | anybody tested using two asterisks, same cdr database? :p |
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