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| 01:56.34 | WIMPy | What's the record for staying on the phone without pause? |
| 02:32.00 | ChannelZ | I was once on a call for 2long |
| 02:33.41 | [TK]D-Fender | Hah, I've gone 3long ..... |
| 02:33.47 | [TK]D-Fender | <- winnar |
| 02:34.26 | WIMPy | I'm currently in the 11th hour :-( |
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| 02:42.13 | ChannelZ | "You hang up!" "No, you hang up! *giggles*" "No, YOU hang up!" |
| 02:43.13 | WIMPy | No, it's not a GF. |
| 02:44.13 | WIMPy | 11:35 |
| 02:44.29 | WIMPy | needs some ice for his ears. |
| 02:44.53 | WIMPy | Headsets with integrated coolling! |
| 02:48.02 | KNERD | it's winter.....heating |
| 02:48.45 | rkeene | Speakerphone. |
| 02:51.45 | WIMPy | Can we do the new years thing again, please? |
| 02:51.53 | WIMPy | I don;t like how this one started. |
| 02:52.36 | KNERD | yes...in 364 and 1.4 days from now |
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| 05:09.45 | carrar | looks around |
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| 05:13.57 | cvance | Hello. I am using pbx in a flash, and I am having some android issues. I am using the native sip support in android, but at times the phone seems to time out (due to it being on a 3g or 4g connection). The device pops offline rather often, and I am unable to call it's extension. How might I address this issue? |
| 05:14.28 | *** join/#asterisk voipboi (~chatzilla@gateway/tor-sasl/tronco) |
| 05:15.35 | voipboi | Does anyone know if you can receive SMS directly from Vitelity into Asterisk? |
| 05:15.56 | [TK]D-Fender | cvance: Disable Qualify on the extension. Change the SIP connection to TCP from UDP. |
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| 05:16.43 | cvance | [TK]D-Fender: Will try, thanks. |
| 05:18.22 | cvance | [TK]D-Fender in the event that the device actually does become unreachable, what happens since qualify is off? How long does the server attempt to connect the endpoint before it pops off to voicemail? |
| 05:21.43 | [TK]D-Fender | It becomes unreachable when it fails to respond. |
| 05:22.07 | [TK]D-Fender | By disabling Qualify * stops ASKING if it's there to "check in" |
| 05:22.38 | [TK]D-Fender | Also increase the registration frequency and enable keep-alives on the client side |
| 05:25.24 | cvance | [TK]D-Fender: Thank's for all the info. I was wondering how to make sip calling through my phone more reliable. I hope this solves it. |
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| 06:08.38 | jpsharp | voipboi: Yes, with asterisk 10 or higher. |
| 06:09.20 | voipboi | jpsharp: i've been looking for documentation; do you have any? |
| 06:10.24 | jpsharp | not handy |
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| 06:31.39 | n2tech | good morning all |
| 06:43.46 | ChannelZ | voipboi: https://wiki.asterisk.org/wiki/display/AST/New+in+10#Newin10-TextMessaging |
| 06:44.01 | voipboi | thankss. |
| 06:44.22 | ChannelZ | the important bit is the new options in sip.conf - then it's a matter of your dialplan to handle incoming |
| 06:44.46 | voipboi | k |
| 06:45.11 | ChannelZ | which I believe you use the MESSAGE() function to read |
| 06:47.58 | ChannelZ | or.. rather than sip.conf, it seems Vitelity is using XMPP (I just logged into my account and saw it) |
| 06:48.28 | voipboi | k. |
| 06:49.11 | voipboi | i'm in my account and can't find xmpp details |
| 06:49.17 | voipboi | just telling me to download their client |
| 06:49.34 | ChannelZ | http://s.ms/ |
| 06:50.58 | ChannelZ | click the 'IM Client Support' link and then look at the instructions for say Pidgin |
| 06:51.22 | ChannelZ | looks like their server is s.ms and you use your DID for a username and whatever password you chose when you activated |
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| 07:09.49 | morfin | hello |
| 07:10.36 | morfin | if i call with callfile and then do Dial in context will they connect? |
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| 07:13.46 | morfin | ? |
| 07:13.56 | morfin | nevermind |
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| 07:25.22 | bulkorok | hi |
| 07:28.01 | ChannelZ | voipboi: hmm got it working here. tip: make sure to use YOURNUMBER@s.ms as your username rather than just your number |
| 07:29.35 | ChannelZ | I'm not sure about Vitelity though, they seem to keep saying "your first 100 SMS are free" but it's still showing/counting up $.01/message |
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| 13:39.22 | davlefouAMD | hi, is it possible to have serval data base with asterisk? |
| 13:42.02 | leifmadsen | davlefouAMD: yes |
| 13:42.05 | leifmadsen | use odbc |
| 13:42.40 | leifmadsen | although your question is vague, so depending on what you're expecting, it may or may not be possible |
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| 13:47.03 | *** join/#asterisk Assid (~assid@unaffiliated/assid) |
| 13:47.08 | Assid | heya |
| 13:48.17 | Assid | probably one of those questions asked a million times.. i used to use termination services from voipjet.. which has shut down; was wondering if anyone could recommend any other service which is similar service & pricing |
| 13:49.15 | leifmadsen | Assid: US or CA? |
| 13:49.33 | leifmadsen | ~voiplist-us |
| 13:49.41 | leifmadsen | hmmm... I forget that tag |
| 13:49.42 | Assid | US;and other countries like india; hongkong ; singapore etc |
| 13:49.56 | leifmadsen | I'm in Canada, but like voip.ms (although flakey recently for some reason) and Unlimitel |
| 13:50.11 | leifmadsen | flowroute is pretty good too |
| 13:50.17 | Assid | nah.. my usage is quite limited.. its cheaper to go pay as you go |
| 13:50.18 | leifmadsen | those are by goto's |
| 13:50.27 | leifmadsen | those are pay as you go |
| 13:50.32 | Assid | yeah flowroute is on my shortlist |
| 13:50.46 | leifmadsen | I don't use any unlimited plans |
| 13:50.46 | [TK]D-Fender | servals are beautiful cats.... |
| 13:51.06 | [TK]D-Fender | My sister had 2. |
| 13:51.16 | [TK]D-Fender | But I've never seen them work with Asterisk before.... |
| 13:51.25 | Assid | [TK]D-Fender: arent they wild ? |
| 13:51.39 | [TK]D-Fender | Assid, debateably domesticated. |
| 13:51.51 | Assid | work with asterisk ??? |
| 13:52.04 | [TK]D-Fender | Assid, many cities won't allow them, but they are often bred in captivity |
| 13:52.46 | [TK]D-Fender | Assid, My sister lived in a large city which "frowned" on the cats... she later moved to the middle of nowhere to keep them. |
| 13:53.12 | Assid | leifmadsen: any idea on the difference of the usage types ? |
| 13:53.23 | leifmadsen | what do you mean? |
| 13:53.42 | Assid | individual /business/business reseller and so on |
| 13:53.49 | Assid | is the DID locked? |
| 13:53.59 | leifmadsen | I'm not the provider, so you'll need to ask them |
| 13:54.08 | leifmadsen | but in north america number porting is law |
| 13:54.13 | davlefouAMD | leifmadsen, i have an data base for sip and iax an another one for the extensions. |
| 13:54.20 | leifmadsen | davlefouAMD: that's fine |
| 13:54.22 | Assid | no i meant the callerid did |
| 13:54.32 | leifmadsen | Assid: again, varies from provider to provider, so check with them |
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| 14:00.43 | Assid | bah.. account suspended |
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| 14:23.18 | *** join/#asterisk twitchnln (~twitch@adsl-184-36-51-168.asm.bellsouth.net) |
| 14:24.18 | twitchnln | Good morning everyone, I was wondering if anyone had ever tried setting cisco 7960/7940 to monitor mailbox other than default? |
| 14:24.23 | *** join/#asterisk hugo (~hugo@unaffiliated/hugo) |
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| 14:32.28 | [TK]D-Fender | twitchnln, You sert it in the peer, not the phone. |
| 14:32.38 | [TK]D-Fender | twitchnln, Do however many you want |
| 14:33.33 | twitchnln | TK: that would work, but the extension has it's own vmb as well, I just need the handset to monitor the general vmb, so they have a blinkie when there is new vm |
| 14:34.14 | [TK]D-Fender | It'll blink for both. If the phone has any kind of brains at all it'll tell which it's blinking for and how many messasges |
| 14:34.26 | twitchnln | I can make it work on Linksys/Polycom/Snom/Grandstream, stands to reason that a 7900 series should be able to do it as well |
| 14:35.43 | twitchnln | can I have multiple mailboxes under the peer? |
| 14:37.33 | *** join/#asterisk Assid (~assid@unaffiliated/assid) |
| 14:37.50 | [TK]D-Fender | <[TK]D-Fender> twitchnln, You sert it in the peer, not the phone. |
| 14:37.50 | [TK]D-Fender | <[TK]D-Fender> twitchnln, Do however many you want |
| 14:38.51 | kaldemar | twitchnln: the sample sip.conf says ";mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes" |
| 14:39.28 | twitchnln | cool, wasn't aware that I could stack them in there, thanks |
| 14:39.32 | *** join/#asterisk mobile_gordita (~Robert@66-87-94-172.pools.spcsdns.net) |
| 14:41.38 | hugo | loader.c: Error loading module 'res_rtp_asterisk.so': /usr/pkg/lib/asterisk/modules/res_rtp_asterisk.so: Undefined PLT symbol "DTLSv1_method" (symnum = 188) <-- on netbsd-6 .. thoughts? |
| 14:41.55 | *** join/#asterisk hurdman (~ygcheny@r2d2.r0b0t.fr) |
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| 14:42.19 | hurdman | hi, is the asterisk-addons-1.6.2.4 ok for a fresh install of asterisk 11 ? |
| 14:43.36 | [TK]D-Fender | twitchnln, or multiple statements |
| 14:43.52 | [TK]D-Fender | hurdman, Obviously not |
| 14:44.12 | [TK]D-Fender | hurdman, Do you 1969 Mustang parts into your brand new 2013 Mustang? |
| 14:44.13 | hurdman | [TK]D-Fender: ok, i was in doubt because the tar.gz date was nov 2012 ^^ |
| 14:44.29 | twitchnln | TK: Thanks for always having the answer I need. |
| 14:44.37 | [TK]D-Fender | hurdman, Look at the date on all the files.... everything got reset |
| 14:44.41 | hurdman | [TK]D-Fender: there's no more asterisk-addons now ? |
| 14:44.45 | hurdman | ah ok |
| 14:44.48 | kaldemar | hurdman: no. you don't need a separate addons package for any branch newer than 1.6.2. |
| 14:45.12 | hurdman | yeah great, thx |
| 14:45.17 | hugo | ok maybe I am approaching this the wrong way - is res_rtp_asterisk.so essential for (s)rtp sessions? |
| 14:45.39 | kaldemar | it is the rtp engine, so yes. |
| 14:45.49 | hugo | thought so |
| 14:45.58 | hugo | any idea about the undefined symbol ? |
| 14:48.45 | *** part/#asterisk Assid (~assid@unaffiliated/assid) |
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| 14:52.18 | *** join/#asterisk StaRetji (~LittleAll@91.142.129.1) |
| 14:52.27 | StaRetji | Howdy folks |
| 14:54.07 | *** join/#asterisk qakhan (~qakhan@50-200-52-14-static.hfc.comcastbusiness.net) |
| 14:54.08 | StaRetji | I need to read callerid and write to a file. It should be something like 555,1,AGI(WriteFile, ${CALLERIDNUMP}) |
| 14:54.21 | StaRetji | but this doesn't work for me, any tip is highly appreciated ;0 |
| 14:54.58 | qakhan | can anyone tell me about Push to Talk in asterisk |
| 14:57.33 | kaldemar | StaRetji: check the contents of the CALLERIDNUMP variable, what your agi does and don't insert a space in front of the argument. |
| 14:57.50 | kaldemar | StaRetji: btw, why are you using agi for that? |
| 14:58.50 | StaRetji | kaldemar: thx dude, I need to send sms to my mobile phone on each call to my sip number |
| 14:59.22 | kaldemar | StaRetji: a simple Set(FILE(/path/to/file)=${CALLERID(num)}) would write caller id number to a file. |
| 14:59.54 | kaldemar | StaRetji: what has writing caller id to a file to do with that? |
| 15:00.53 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:00.53 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 15:01.19 | [TK]D-Fender | If you're in an AGI ... it laready HAS access to the callerID. And antyhing else it feels like grabbing |
| 15:01.29 | StaRetji | kaldemar: sorry for the delay, mum called lol, she is hard on hangup :D |
| 15:01.33 | [TK]D-Fender | Someone has completely missed the point of what AGI is.... |
| 15:01.54 | StaRetji | I just need option where I write caller id to a file |
| 15:02.08 | StaRetji | I have script whish sends sms already |
| 15:02.13 | kaldemar | is that the only thing the agi does? |
| 15:02.19 | StaRetji | but I get empty sms, like you missed call from |
| 15:02.23 | StaRetji | and nothing :) |
| 15:02.53 | StaRetji | I will try example from above |
| 15:03.04 | kaldemar | and why are you writing the clid to a file? why are you not passing it to the script that sends the message? |
| 15:03.31 | StaRetji | well, script is very simple, it reads from WhoCalled.txt |
| 15:03.39 | StaRetji | and ads it to sms subject |
| 15:04.02 | kaldemar | if you can give it arguments, using a file would make no sense. |
| 15:04.51 | StaRetji | well, I bet you are right, but I am noob so I play with what I have |
| 15:05.08 | StaRetji | I don't know how to send arguments to external script |
| 15:05.12 | StaRetji | so, I do it simple |
| 15:05.15 | StaRetji | write to a file |
| 15:05.18 | StaRetji | read from a file |
| 15:05.35 | kaldemar | writing to a file and reading it is far from simple. how are you calling the sms script? |
| 15:05.51 | StaRetji | AGI (SendSMS) |
| 15:06.08 | StaRetji | and SendSMS is bash script |
| 15:06.16 | kaldemar | and why are you using agi? |
| 15:06.22 | StaRetji | well |
| 15:06.25 | StaRetji | I don't know lol |
| 15:06.31 | [TK]D-Fender | If it's an AGI you don't need to PASS anything |
| 15:06.35 | StaRetji | I guessed if someone calls my 555 number |
| 15:06.35 | [TK]D-Fender | AGI pulls whatever you tell it to |
| 15:06.39 | [TK]D-Fender | Go read the book. |
| 15:06.41 | StaRetji | so, I did 555,1,blabla |
| 15:06.43 | [TK]D-Fender | Variables are variables... |
| 15:06.50 | StaRetji | 555,2,writefile |
| 15:06.58 | StaRetji | 555,3, sendsms |
| 15:07.07 | [TK]D-Fender | StaRetji, AGI has full access to all your channel variables, callerID, etc. |
| 15:07.19 | [TK]D-Fender | StaRetji, And cal do ALL of that all in one. |
| 15:07.22 | [TK]D-Fender | can* |
| 15:07.31 | kaldemar | then don't. System(/path/to/SendSMS ${CALLERID(num)}) <-- you'll have the caller id in $1 in your script. |
| 15:08.11 | [TK]D-Fender | That presumes that its acting as an AGI was a lie and that he is also abusing that tool.... |
| 15:08.29 | kaldemar | using agi and writing stuff to a file is about the most complicated way of doing what you need. |
| 15:09.12 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:10.47 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
| 15:11.02 | qakhan | can anyone tell me about Push to Talk in asterisk |
| 15:12.13 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 15:20.46 | *** join/#asterisk _Corey_ (~chatzilla@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 15:24.09 | StaRetji | kaldemar: Set(FILE(/path/to/file)=${CALLERID(num)}) wrote this to a file: ${CALLERID(num)}) |
| 15:24.29 | StaRetji | or I am doing something wrong |
| 15:26.38 | kaldemar | show CLI output for that |
| 15:31.50 | StaRetji | kaldemar: http://pastebin.com/LxRs8R2N thx |
| 15:34.02 | [TK]D-Fender | StaRetji, /agi-bin/WhoCalled.txt)=$CALLERID(num)}" |
| 15:34.10 | [TK]D-Fender | staYou are clearly missing the { after the $ |
| 15:34.21 | [TK]D-Fender | StaRetji, Pay attention to your syntax |
| 15:35.05 | *** join/#asterisk Romeo- (~romi@unaffiliated/romeo/x-000000001) |
| 15:35.07 | StaRetji | ahh, sorry for that |
| 15:35.23 | StaRetji | was having tough last few days lol |
| 15:37.11 | kaldemar | and you really should not use a file for that. |
| 15:38.42 | StaRetji | ok, I will try do it properly then, thx |
| 15:42.38 | *** join/#asterisk blizzow1 (~jburns@67.50.165.58) |
| 15:49.43 | *** join/#asterisk moos3 (~moos3@pool-72-73-84-223.ptldme.east.myfairpoint.net) |
| 16:24.07 | qakhan | Please tell me about Push to Talk in asterisk |
| 16:31.09 | *** join/#asterisk gusto (~gusto@adsl-dyn-75.95-102-92.t-com.sk) |
| 16:33.38 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 16:40.55 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 16:43.04 | *** join/#asterisk arag00rn (~arag00rn@trapdoor.smallunix.net) |
| 16:45.16 | arag00rn | Hi all, I've a strange issue. I'm getting MOH vs ringing on outbound calls. I'm using asterik 1.8.19 with freepbx |
| 16:45.29 | arag00rn | do you have any idea? |
| 16:50.29 | *** join/#asterisk cbdev (~fnord@hieristdas.internetzuen.de) |
| 16:52.17 | *** part/#asterisk ocrete (~ocrete@gentoo/developer/tester) |
| 17:07.48 | WIMPy | qakhan: That's not an Asterisk thing. It's a terminal thing. |
| 17:09.06 | qakhan | WIMPy can you help me to understand what is terminal? |
| 17:09.31 | WIMPy | A phone |
| 17:09.38 | qakhan | Thanks |
| 17:10.57 | qakhan | is there any possibility of PTT in asterisk? |
| 17:11.21 | WIMPy | Asterisk doesn't care. |
| 17:11.27 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:12.36 | WIMPy | Depending on your terminals you may want to an alert-info header. But apart from that it's up to the terminals you (want to) use. |
| 17:15.09 | *** part/#asterisk twitchnln (~twitch@adsl-184-36-51-168.asm.bellsouth.net) |
| 17:18.35 | moos3 | question is IAX2 not valid anymore ? |
| 17:18.45 | moos3 | in the means of channels |
| 17:18.55 | WIMPy | What makes you think so? |
| 17:20.29 | *** join/#asterisk Leddy (leddy@krypton.evosurge.com) |
| 17:25.40 | moos3 | app_dial.c:1780 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown) |
| 17:26.02 | WIMPy | Something that is not registered? |
| 17:28.55 | *** join/#asterisk _Corey_ (~chatzilla@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 17:37.34 | kaldemar | moos3: you don't have chan_iax2.so loaded for some reason. |
| 17:38.00 | moos3 | but calls to them work just fine which is odd |
| 17:38.20 | kaldemar | "to them"? |
| 17:38.46 | moos3 | yea |
| 17:38.55 | moos3 | I have 4 iax extension |
| 17:39.07 | moos3 | all the calls work but they all throw those errors |
| 17:39.23 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.229) |
| 17:39.36 | kaldemar | i highly doubt that they work. pastebin what you see in CLI with verbosity enabled. |
| 17:39.41 | WIMPy | So one ofe those 4 isn't reachable. |
| 17:39.53 | WIMPy | -e |
| 17:40.33 | *** join/#asterisk ffs (~garland@unaffiliated/ffs) |
| 17:40.40 | kaldemar | that's the case. cause 20 slipped my eye. |
| 17:40.55 | moos3 | k |
| 17:41.03 | kaldemar | asterisk does not have an address for a destination. |
| 17:41.51 | WIMPy | Not sensible to make that generate cause 20, but I think that's what happens. |
| 17:42.06 | moos3 | yea |
| 17:43.30 | *** join/#asterisk acedia (~rage@unaffiliated/ffs) |
| 17:43.44 | *** join/#asterisk elico (~Thunderbi@109.67.204.3) |
| 17:43.46 | [TK]D-Fender | <kaldemar> moos3: you don't have chan_iax2.so loaded for some reason. <- no, otherwise it get "channel type not implemented" |
| 17:44.18 | [TK]D-Fender | .... and not I catch up .. to other people catching up |
| 17:44.20 | [TK]D-Fender | now* |
| 17:45.01 | moos3 | it was loaded |
| 17:45.12 | moos3 | i think one of the devices lost its registration |
| 17:46.18 | [TK]D-Fender | that's the usualy case |
| 17:51.18 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:51.18 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:53.24 | *** join/#asterisk juicec (1839c48d@gateway/web/freenode/ip.24.57.196.141) |
| 17:59.22 | juicec | Good afternoon all :D |
| 17:59.37 | juicec | I'm trying to perform a migration from data center to another and am having issues with licensing |
| 17:59.53 | juicec | Is there someone I can talk to about putting my new servers host ids onto the licenses as a type of "trial" ? |
| 18:01.53 | _Corey_ | juicec: What kind of licenses are you talking about? |
| 18:02.55 | juicec | g729 + fax + a skype plugin |
| 18:03.14 | *** join/#asterisk imox (~imox@91-66-32-57-dynip.superkabel.de) |
| 18:05.22 | WIMPy | You should contact Digium support. |
| 18:08.01 | juicec | Will do |
| 18:13.59 | qakhan | is there any option in meetme to mute all users when they join a conf room, if some one want to speak he/she press any button to speak? |
| 18:14.52 | *** join/#asterisk gg608f (~Adium@187.207.6.250) |
| 18:17.26 | [TK]D-Fender | ~book |
| 18:17.26 | 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 |
| 18:17.30 | [TK]D-Fender | qakhan, ^^^^ |
| 18:18.02 | *** join/#asterisk dpilon (~dpilon@50.138.200.155) |
| 18:18.44 | qakhan | [TK]D-Fender i didnt find my answer in this book |
| 18:19.03 | [TK]D-Fender | "core show application meetme" |
| 18:19.08 | [TK]D-Fender | Read the sample config |
| 18:19.10 | [TK]D-Fender | The options are there |
| 18:20.53 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 18:22.35 | *** join/#asterisk Russ (~russ@pool-74-100-57-85.lsanca.fios.verizon.net) |
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| 18:43.56 | qakhan | [TK]D-Fender i used 'm' option in meetme app |
| 18:44.17 | qakhan | but i cant speck while i press any any |
| 18:44.20 | qakhan | key* |
| 18:48.41 | *** join/#asterisk nicknam1232 (021f9725@gateway/web/freenode/ip.2.31.151.37) |
| 18:50.26 | *** join/#asterisk nicknam1232 (021f9725@gateway/web/freenode/ip.2.31.151.37) |
| 18:53.17 | *** join/#asterisk navaismo (~navaismo@189.144.207.195) |
| 18:58.25 | qakhan | ? |
| 18:59.07 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 19:02.15 | Katty | oh man i am being bad today. cheese its AND mt dew for snack. |
| 19:02.35 | Katty | don't tell the calorie gestapo. ssh. |
| 19:02.46 | Penguin | Sounds like I'm going to have to give someone a spankin'. |
| 19:02.50 | [TK]D-Fender | qakhan, If you put people in muted... then they can't speak.\ |
| 19:02.52 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 19:03.35 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 19:04.50 | Katty | dear pepsico company, why are you putting brominated vegetable oil in my mt dew? |
| 19:05.15 | [TK]D-Fender | Because non-brominated vegetable oil would be just WRONG..... |
| 19:05.32 | Katty | i t hink we can leave all kinds of vegetable oil out of my drink. |
| 19:05.50 | Katty | brominated or not. |
| 19:05.58 | nicknam1232 | Hope you are all well, good. Cool. |
| 19:06.06 | qakhan | [TK]D-Fender my question is people should be in mute state but they can speak by pressing any key like # or 0 etc... |
| 19:06.28 | [TK]D-Fender | Katty, It's an emulsifier. I'm sure you wouldn't want your Mt. Dew to ... glomp |
| 19:06.46 | [TK]D-Fender | qakhan, Where do you see anything about using a key to enable them to speak? |
| 19:07.22 | Katty | whatever is causing it to glomp. we don't need that in there either. |
| 19:07.51 | qakhan | [TK]D-Fender i didnt get you |
| 19:08.20 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 19:08.20 | [TK]D-Fender | qakhan, What don't you get? You asked about using keypresses to unmute. Do YOU see an option that lets them do that? |
| 19:08.43 | Katty | i should become a chemist. |
| 19:08.48 | Katty | then i can mix up my own sodas. |
| 19:08.53 | Katty | without glomping. |
| 19:09.23 | [TK]D-Fender | Katty, The fact you're putting solids into a liquid and expecting them to diffuse is the challenge. Hence the emulsifier. If you don't want that problem .... drink water ;) |
| 19:10.00 | Katty | i'll diffuse you in water in a minute. |
| 19:10.03 | qakhan | [TK]D-Fender no i dont see any thing in meetme app |
| 19:10.19 | Katty | just as soon as jesus turns water into mt dew. |
| 19:11.23 | Katty | that probably wasn't very nice. |
| 19:11.28 | Katty | no offense meant. |
| 19:12.20 | [TK]D-Fender | qakhan, That's because it ISN'T an option. |
| 19:12.52 | [TK]D-Fender | Katty, Jesus walking on water wouldn't be a miracle ... just disgusting ;) |
| 19:13.31 | qakhan | [TK]D-Fender is there any way? |
| 19:15.25 | [TK]D-Fender | qakhan, Obviously nothing to do with app_meetme\ |
| 19:30.47 | qakhan | any other solution for this |
| 19:30.49 | ChannelZ | qakhan: switch to ConfBridge |
| 19:30.58 | ChannelZ | depending on your version of Asterisk |
| 19:31.31 | qakhan | ChannelZ i am using 1.4.38 |
| 19:32.11 | [TK]D-Fender | Upgrade |
| 19:32.13 | [TK]D-Fender | 1.4 is DEAD |
| 19:33.04 | qakhan | currently i cannt, |
| 19:33.16 | ChannelZ | then you are stuck |
| 19:33.29 | qakhan | awwwww |
| 19:34.28 | [TK]D-Fender | Over 2 years old on a branch that has been functionally dead longer still... |
| 19:35.08 | ChannelZ | Why can't you upgrade? |
| 19:35.42 | qakhan | i am using agentcallbacklogin app in 1.4.38 |
| 19:35.49 | [TK]D-Fender | So stop |
| 19:36.05 | qakhan | this is not available in new version |
| 19:36.08 | [TK]D-Fender | That isn't a "reason" |
| 19:36.10 | [TK]D-Fender | You don't NEED it |
| 19:36.20 | [TK]D-Fender | You have clearly not read how AddQuememeber works |
| 19:36.30 | [TK]D-Fender | AddQueueMember * |
| 19:36.53 | [TK]D-Fender | Any other "reasons"? |
| 19:37.22 | qakhan | [TK]D-Fender yes you are right |
| 19:37.34 | qakhan | but it is not that easy |
| 19:37.41 | [TK]D-Fender | qakhan, Isn't it? |
| 19:37.54 | [TK]D-Fender | qakhan, You haven't given us any reason to think otherwise |
| 19:38.09 | qakhan | Sir i told you |
| 19:38.22 | qakhan | we have integrade our software with ASterisk |
| 19:38.31 | qakhan | using agentcallbacklogin |
| 19:38.44 | [TK]D-Fender | qakhan, All I've seen is your asking questions about a dead branhc for over a year and having a clear problem reading documentation for yourself. |
| 19:38.49 | [TK]D-Fender | qaCHANGE your software then |
| 19:39.06 | qakhan | hmmm ok |
| 19:39.07 | qakhan | thanks |
| 19:40.10 | qakhan | in which version of asterisk confbridge is available |
| 19:40.34 | [TK]D-Fender | qakhan, Do not waste time thinking about anything lower than 1.8 |
| 19:40.50 | qakhan | hmm ok |
| 19:40.50 | [TK]D-Fender | qakhan, Even then, 1.8 hits ELO at the end of this year |
| 19:40.52 | [TK]D-Fender | EOL* |
| 19:41.10 | [TK]D-Fender | qakhan, Save yourself further grief and plan towards * 11 |
| 19:41.20 | qakhan | hmmm |
| 19:41.53 | mjordan | [TK]D-Fender: 1.8 should hit security fix only 2014-10-21 |
| 19:42.00 | mjordan | so there's still a bit of life in it yet ;-) |
| 19:42.32 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 19:42.52 | [TK]D-Fender | mjordan, Might have mixed up * 10 on that one... |
| 19:43.11 | [TK]D-Fender | mjordan, Still, I don't want him coming back here again that soon with the same insanity :p |
| 19:43.41 | mjordan | And I'm also a fan of as many people going to 11 too :-D |
| 19:44.11 | ChannelZ | did ConfBridge show up in 1.6 or 1.8? I don't remember but I thought the first iteration was really stripped |
| 19:44.22 | [TK]D-Fender | mjordan, So far I haven't heard any horror stories more than you get for any release period. That in mind better to just "git 'er done" |
| 19:44.42 | mjordan | It is in 1.8, but it was overhauled significantly in Asterisk 10+. |
| 19:44.46 | mjordan | [TK]D-Fender: agreed |
| 19:44.55 | mjordan | ChannelZ: ^^^ |
| 19:45.10 | mjordan | I would say use MeetMe in 1.8, honestly. |
| 19:45.31 | [TK]D-Fender | Once FreePBX has a stable release supporting * 11 I'll be migrating my office to it as well. |
| 19:46.02 | [TK]D-Fender | mjordan, ConfBridge looks to basically start handing MeetMe its own ass in 11.... |
| 19:46.45 | mjordan | that was the goal! |
| 19:46.50 | Micc | TK, I never saw you as a FreePBX guy. |
| 19:47.48 | [TK]D-Fender | Micc, it's for my office. I was on a closed commercial GUI from 2005/10 through 2012/08 |
| 19:48.20 | [TK]D-Fender | Micc, I got them off it for ease of support, cost, and realistic requirements reasons |
| 19:49.08 | [TK]D-Fender | Micc, if I didn't care about the supportability by others, then hell yeah, I'd do it myself.... |
| 19:49.33 | [TK]D-Fender | Micc, I am the IT dept here, and head office has little Linux experience. |
| 19:51.15 | leifmadsen | mjordan: +1 |
| 19:52.21 | [TK]D-Fender | mjordan, * 11 is indeed major in my eyes. Adds LTS including T.38 termination, SRTP, ConfBridge++, and a pile of other goodies. Real world stuff that people care about. |
| 19:53.05 | [TK]D-Fender | mjordan, And as soon as * has that "stable" feeling about it, absolutely everything under it is on the chopping block.... |
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| 20:00.54 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 20:33.31 | *** join/#asterisk schultza (~schultza@rc1.rcherbals.com) |
| 20:34.54 | schultza | what is the recommended (linux) disk space sizes for asterisk... im building a test environment, but would still like to know production environment sizes... i cant seem to find this info in "the definitive guide 2011" book |
| 20:40.40 | leifmadsen | well, it depends entirely what you're doing with asterisk |
| 20:40.52 | leifmadsen | if you're not recording anything, then disk space is minimal |
| 20:41.21 | leifmadsen | that being said... I've never had a problem with 15-20GB with the grow option enabled for a dev environment |
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| 20:41.34 | leifmadsen | actual disk space usually ends up being mostly the OS anyways |
| 20:44.23 | wdoekes | as long as you remember to rotate your debug/verbose logs |
| 20:44.38 | leifmadsen | +1 |
| 20:45.05 | leifmadsen | although I use vagrant + chef, so my virtual machines never stay built for more than a couple weeks |
| 20:45.17 | leifmadsen | vagrant destroy --force asterisk && vagrant up asterisk |
| 20:45.23 | leifmadsen | do that for each project |
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| 21:05.50 | schultza | vagrant and chef? never heard of those |
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| 21:20.37 | dijib | sip_write: Can't send 10 type frames with SIP write |
| 21:20.43 | dijib | anybody seen this one? |
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| 21:30.02 | leifmadsen | schultza: then you're missing out :) |
| 21:31.08 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.1.1 (2013/01/02), 10.11.1 (2013/01/02), 1.8.19.1 (2013/01/02), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 21:39.14 | rrittgarn | Anybody do much with res_corosync? Can't seem to get it to load the module even after re-compiling and it showing up in menuselect. |
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| 21:50.57 | schultza | leifmadsen, is there a recommended source to get started on those resources (pdf, etc)? |
| 21:51.14 | leifmadsen | just the google |
| 21:51.29 | leifmadsen | rrittgarn: more info required I believe |
| 21:52.03 | *** join/#asterisk RSNETSAU (~RS.Net@ppp203-122-217-95.static.internode.on.net) |
| 21:53.00 | schultza | ok |
| 21:53.07 | RSNETSAU | Hey guys need assistance if anyboody can help it would be appreciated: http://forums.whirlpool.net.au/forum-replies.cfm?t=2034589&#bottom |
| 21:56.57 | rrittgarn | lef: I have two vms, one i can get corosync to start, and then manually module load res_corosync.so, and not on the other, so I'm thinking there's an issue with my config / that box... so I'll keep this problem out of this channel for now... |
| 21:57.25 | rrittgarn | leif: Unless you happen to know what the heck API error 6 is... |
| 21:59.27 | *** join/#asterisk elico (~Thunderbi@109.67.204.3) |
| 22:07.25 | justdave | would a Polycom phone consider it a network topology change if it renews a DHCP lease and the lease ACK comes from a different server than it did before (even if the details of the lease itself are still the same) ? |
| 22:07.55 | justdave | common ocurrence in isc-dhcpd with a multi-server failover configuration for DHCP |
| 22:08.43 | justdave | phones seem to be rebooting at random when their DHCP lease renews |
| 22:11.41 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 22:12.06 | jacekowski | to be honest, i don't understand why phones reboot at all when network config changes |
| 22:12.23 | jacekowski | considering that most of those phones run linux and doing it without reboot would be easy |
| 22:12.32 | justdave | yeah, no kidding. :) |
| 22:15.39 | *** join/#asterisk war9407 (war@c-71-62-63-105.hsd1.va.comcast.net) |
| 22:17.59 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 22:19.51 | rrittgarn | Leif: with corosync, it only works with multicast right? So I have to make sure that if I'm using asterisk/corosync across facilities that I tunnel the multi-cast trafic right? |
| 22:20.07 | schultza | thanks leif for the info (vagrant and chef)... and it works with my current choice of vm software. |
| 22:21.24 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 22:25.16 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 22:29.29 | *** join/#asterisk reisi (koivunej@mustatilhi.cs.tut.fi) |
| 22:30.48 | reisi | with 1.8, what is wrong with this dialplan line: same => n,GotoIf($[${paged} != ""]?skipPaged:doPaged), error is on !=: syntax error: syntax error, unexpected '!=', expecting $end; |
| 22:31.41 | Qwell | reisi: Your paged variable is empty. |
| 22:31.50 | Qwell | GotoIf($[ != ""]?...) |
| 22:31.57 | Qwell | (hint: quote it) |
| 22:32.26 | reisi | hmmm strange, it really should not be |
| 22:33.04 | reisi | oh actually yes, it's empty string, but should be set |
| 22:34.17 | kaldemar | "" is not an empty string but two quotes. |
| 22:34.43 | reisi | yes, quoting it makes it work |
| 22:35.02 | leifmadsen | you could have put an x or whatever too |
| 22:35.22 | leifmadsen | GotoIf($[x${foo} != x]?error_handler,1) |
| 22:35.28 | Qwell | GotoIf($[leifmadsenisanub${paged} != leifmadsenisanub]?...) |
| 22:35.34 | leifmadsen | totally valid! |
| 22:37.09 | reisi | :) |
| 22:47.08 | reisi | does application Hangup without any arguments always exit non-zero? |
| 22:51.33 | KNERD | Who has some awesome ideas on getting through Carlo Sime's Great VOIP wall? |
| 22:56.03 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 23:00.30 | saint_ | Greetings all. So I saw the security update in the mailing list. I did install my asterisk following the asteriskdocs guide (using svn). When I do core show version it says Asterisk SVN-branch-1.8-r378164 . How do I know if it's 1.8.11 cert or 1.8.19.1 , and how do i upgrade it to the latest stable / secure version ? |
| 23:01.22 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:02.29 | mjordan | saint_: you downloaded a particular snapshot of an Asterisk branch, so it isn't a version. Versions are tags made from those branches. |
| 23:03.20 | mjordan | saint_: in you particular case, you know it's the 1.8 branch because it's "SVN-branch-1.8", as opposed to "SVN-branch-1.8.11". That means 1.8.19.1 was made from that branch (actually, it was made from 1.8.19.0, which was made from that branch) |
| 23:04.00 | *** join/#asterisk MrRobert (cf4322ee@gateway/web/freenode/ip.207.67.34.238) |
| 23:04.14 | MrRobert | hey guys, anyone know why i can't use the Agents command via AMI? It says Invalid/unknown command: Agents. Use Action: ListCommands to show available commands. |
| 23:04.44 | MrRobert | The user has system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate options set in manager.conf |
| 23:04.50 | MrRobert | for read and write |
| 23:05.00 | saint_ | mjordan: so does it mean that i need to go to a sub branche or whatever it is called in order to have the latest version, or is it better to get the latest 1.8.x source code, compile it, and install it, without using svn ? |
| 23:05.02 | MrRobert | asterisk 11 |
| 23:05.16 | mjordan | saint_: in your particular case, you didn't get the security vulnerability patches that went into the 1.8 branch, as those were commits 378269 and 378303. Your checkout is from r378164. You should either apply the patches in the AST, or update your checkout. |
| 23:06.08 | saint_ | i tried svn up http://svn.asterisk.org/svn/asterisk/branches/1.8/ but it did not do anything. is there anything else to do ? \ |
| 23:06.22 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 23:07.20 | mjordan | saint_: go into your actual checkout of the 1.8 branch and run 'svn update' |
| 23:07.46 | saint_ | great, that did it |
| 23:07.54 | MrRobert | when i run listcommands, Agents doesn't appear which is to be expected I imagine, but its documented |
| 23:07.57 | saint_ | thanks a ton. I'll recompile and install. |
| 23:08.06 | mjordan | saint_: You may want to read up on subversion if you're going to use it for deploying Asterisk :-) |
| 23:09.55 | saint_ | mjordan: will do. do you have an URL you could recommend ? |
| 23:11.27 | mjordan | saint_: nope. Google is probably your friend there. |
| 23:11.37 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-uycsakfghgvbrerx) |
| 23:11.39 | saint_ | lol.. will do. thanks a ton for the trick. |
| 23:15.17 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:16.07 | MrRobert | any ideas anyone? |
| 23:19.37 | saint_ | damn it, mjordan left. Can someone tell me how would I know that Asterisk SVN-branch-1.8-r378375 is the latest most secure version for 1.8 branch ? |
| 23:21.03 | MrRobert | Please help :) http://pastebin.com/zWjE56eu |
| 23:24.45 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 23:27.16 | MrRobert | heyy TK |
| 23:27.22 | MrRobert | guess what i have another issue lol |
| 23:27.27 | MrRobert | http://pastebin.com/zWjE56eu |
| 23:29.08 | *** join/#asterisk elico (~Thunderbi@109.67.204.3) |
| 23:29.41 | [TK]D-Fender | MrRobert: Show the complere AMI dump |
| 23:35.47 | MrRobert | all it says is that error message, or is there a way to get more information? |
| 23:35.51 | MrRobert | <PROTECTED> |
| 23:36.13 | MrRobert | there are about 30k responses from the live system that don't have any influence |
| 23:39.24 | *** join/#asterisk DropsOfSerenity (~justin@vpn.bctconsulting.com) |
| 23:39.37 | [TK]D-Fender | Where do you see that this is a valid command? |
| 23:40.00 | MrRobert | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_Agents |
| 23:40.28 | DropsOfSerenity | Hi guys, i'm making a callfile with Maxretries set to 3, and i'm trying to catch the case in the dialplan where it tries all 3 times and gets BUSY at the very end. I've tried the "failed" extension but it doesn't seem to be going there. |
| 23:41.30 | [TK]D-Fender | DropsOfSerenity: Show us |
| 23:41.35 | [TK]D-Fender | ~pb |
| 23:41.35 | 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. |
| 23:41.37 | [TK]D-Fender | ^^^ |
| 23:42.15 | [TK]D-Fender | MrRobert: Could have sworn chan_agent was toasted by this point... |
| 23:42.48 | MrRobert | How then can i get a list of agents? It's possible it was left in the docoumentation by mistake |
| 23:42.56 | MrRobert | list of agents and their stati |
| 23:43.05 | [TK]D-Fender | MrRobert: Are yo8u actually using agents.conf? |
| 23:43.33 | MrRobert | nope |
| 23:43.45 | DropsOfSerenity | [TK]D-Fender, http://fpaste.org/64gx/ |
| 23:43.48 | [TK]D-Fender | So what you expecting to get anyway then? |
| 23:44.10 | [TK]D-Fender | DropsOfSerenity: And the call file.... |
| 23:44.47 | *** join/#asterisk Molo (~Molo@c-76-31-26-101.hsd1.tx.comcast.net) |
| 23:44.55 | MrRobert | [TK]D-Fender well we want the status of all people currently connected |
| 23:44.57 | MrRobert | all extensions |
| 23:45.01 | MrRobert | aka agents to us |
| 23:45.17 | DropsOfSerenity | [TK]D-Fender, http://fpaste.org/rbRv/ |
| 23:46.10 | [TK]D-Fender | MrRobert: not extensions. CHANNELS |
| 23:46.24 | [TK]D-Fender | MrRobert: Start with that in mind... |
| 23:47.19 | DropsOfSerenity | [TK]D-Fender, I can get it to drop to the -BUSY extension each retry, but I want to act in my dialplan on when it fails on BUSY the last time. |
| 23:47.43 | [TK]D-Fender | DropsOfSerenity: Re-paste both and include your dialplan and show us where you feel this "catch" attempt of yours is... |
| 23:48.54 | DropsOfSerenity | [TK]D-Fender, http://fpaste.org/vmk4/ |
| 23:49.06 | DropsOfSerenity | the fax-tx context isn't important because it never goes there when the call isn't picked up |
| 23:49.22 | *** join/#asterisk Tecan (~fasdf@unaffiliated/unit41) |
| 23:49.34 | DropsOfSerenity | the commented out bit will catch each BUSY attempt |
| 23:49.57 | Tecan | does anyone here run pbx services across canada ? |
| 23:50.00 | [TK]D-Fender | It should not. |
| 23:50.11 | [TK]D-Fender | <PROTECTED> |
| 23:50.18 | DropsOfSerenity | nope, it works fine |
| 23:50.20 | [TK]D-Fender | Anything after . = waste |
| 23:50.35 | DropsOfSerenity | no, because -BUSY is something asterisk appends |
| 23:50.52 | [TK]D-Fender | it'd fall there no matter what. The thought that the word "busy" on the end means anything is incorrect |
| 23:51.13 | DropsOfSerenity | either way it doesn't matter because that's not what i'm looking for |
| 23:51.14 | [TK]D-Fender | the "." alone captures EVERYTHING from that dot onwards |
| 23:51.48 | DropsOfSerenity | mk, then it was just falling through before |
| 23:51.53 | DropsOfSerenity | but why isn't it going to the failed extension |
| 23:54.20 | [TK]D-Fender | DropsOfSerenity: Because it isn't in the right context |
| 23:54.59 | DropsOfSerenity | what context should it be in, the logs show it failing inside the fax-transmit context |
| 23:55.29 | [TK]D-Fender | that's your Channel:. That has nothing to do with the context defined in your call-file |
| 23:57.20 | DropsOfSerenity | but it's never reaching that context because the channel is never picked up |
| 23:57.56 | [TK]D-Fender | DropsOfSerenity: You did not put the failed exten into the context specified in your call file. |
| 23:59.54 | DropsOfSerenity | hmmm right you are, but now it's going there every time it gets busy, rather than at the very end |