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| 04:47.33 | linocisco | hi all |
| 04:47.51 | linocisco | i can't flash my cisco 7942G into sip well. |
| 04:53.53 | linocisco | my cisco phone keep rebooting and started trying to find to upgrad firmware |
| 04:55.26 | linocisco | i dont know how to make it normal mode |
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| 06:38.36 | help_me_pls | i want to make the extension 72 80 ring on 3 new phones we got... i configured the phones and they work individually with 7281, 7282 and 7283, but when i call 7280 it just answers and then hangs up after 1 second |
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| 06:39.31 | help_me_pls | this is what i added to my extensions.conf, is this wrong? http://pastebin.com/fMEgT2KM |
| 06:40.34 | help_me_pls | do i have to add something more to the extensions.conf to get it to work? |
| 06:40.48 | help_me_pls | could somebody please point me in the right direction |
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| 07:04.33 | kaldemar | help_me_pls: ",5" means "quit dialing after 5 seconds and move on to the next priority" |
| 07:06.07 | kaldemar | show a CLI output for a call with verbosity. |
| 07:09.47 | help_me_pls | http://pastebin.com/grW8AWzC |
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| 07:14.11 | help_me_pls | okay i got it |
| 07:15.02 | help_me_pls | under [incoming-calls] i added this http://pastebin.com/Gth9cHFQ |
| 07:15.36 | help_me_pls | i just looked at what we had there for the other numbers, copied it and change it to 7280 |
| 07:16.14 | kaldemar | stop stuffing answer() into every extension you make. |
| 07:17.25 | kaldemar | in your last paste there is absolutely no need for the answer app, it will just cause confusion in call progress state. |
| 07:18.13 | help_me_pls | hmm... im pretty new to asterisk and got this config from the guy before me.... how should it look correctly? |
| 07:18.53 | kaldemar | remove the answer app. |
| 07:19.32 | help_me_pls | ive just been learning by doing... kinda going along by trial and error (lots of error).... so everything i do is kinda based on what the other guy did when he first made it |
| 07:19.47 | help_me_pls | sooo like this? exten => 7280,n,set(CALLERID(name)=Heraeus) exten => 7280,n,Dial(SIP/281&SIP/282&SIP/283,40,m(inspectron)tT) exten => 7280,n,VoiceMail(7273,su) exten => 7280,n,Hangup() |
| 07:20.04 | help_me_pls | or do you mean under the incoming calls part? |
| 07:21.39 | kaldemar | an extension must always start with priority 1. so you'd have to change the "n" priority of the set to "1". |
| 07:22.01 | kaldemar | the incoming also does not need the answer. |
| 07:23.56 | kaldemar | ~book |
| 07:23.56 | 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 |
| 07:24.52 | kaldemar | some reading on dialplan basics will help you plenty. if you just copy and paste stuff without understanding what they do, it will be a rocky road. |
| 07:31.30 | help_me_pls | ill look at it, thanks man |
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| 07:40.16 | phix | help_me_pls: Interesting nick |
| 07:40.44 | help_me_pls | straight to the point.... i know what i am |
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| 07:46.00 | tparcina | Where can I find information what this ":0:1" mean in ${CALLERID(num):0:1}? |
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| 07:52.36 | phix | GANG! |
| 07:58.46 | tparcina | phix: Do you know where can I find information what this ":0:1" mean in ${CALLERID(num):0:1}? |
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| 08:13.07 | bulkorok | how do I stopp receivefax cleanly!? |
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| 08:13.35 | bulkorok | hangup rquest kills the whole reception... but I suppose only the sender can stop!? |
| 08:15.18 | kaldemar | tparcina: https://wiki.asterisk.org/wiki/display/AST/Manipulating+Variables+Basics |
| 08:15.37 | kaldemar | tparcina: https://wiki.asterisk.org/wiki/display/AST/Selecting+Characters+from+Variables |
| 08:22.59 | tparcina | kaldemar: Thank you. |
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| 09:11.10 | romaintordo | good afternoon, I am looking for a developer expert on FreePBX module development to push any CDR to our CRM, does any of you happen to know a company or a freelancer interested in this kind of project? - thanks |
| 09:11.33 | tparcina | Is this statement correct? In dialplan, labels don't change execution order of dialplan. They are used so that we can easily jump to that priority. And execution order of priority with label is the same like we don't have label at all. |
| 09:19.53 | wdoekes | tparcina: correct. labels don't change the order (how could they?) |
| 09:21.09 | tparcina | wdoekes: Thank you. I though it's better to check. :) |
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| 09:49.25 | RadJackson | Hi everyone, |
| 09:51.15 | RadJackson | Actually we are using Asterisk to make outgoing calls in France, our sip trunk is linked to a french operator, while creating the context we made sure not to add the country calling code to the CallerId , despite that, it is still displaying it. |
| 09:51.47 | RadJackson | Is this linked to our developement process? or may be we should contact the provider? |
| 09:56.09 | *** join/#asterisk mihamina (~mihamina@41.188.46.121) |
| 09:57.46 | RadJackson | Ok thank you very much |
| 09:59.13 | WIMPy | WHERE is it displayed? |
| 09:59.29 | RadJackson | the receiving device screen |
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| 10:02.02 | phix | tparcina: I would say the asterisk documentation would be a great place to start |
| 10:02.14 | phix | bulkorok: killall -9 asterisk :P |
| 10:02.23 | WIMPy | Usually you don't have any influence on how it's displayed at the caller. Or is it only for callees with the same provider? |
| 10:02.25 | phix | that will kill it for sure |
| 10:02.33 | phix | hai WIMPy! |
| 10:03.03 | WIMPy | Hi phix |
| 10:03.06 | bulkorok | phix: sure... but 'hangup request channel...' makes it too :-) |
| 10:03.11 | phix | :P |
| 10:03.36 | phix | heh yeah, my example was like using a nuke to kill a fly instead of just using a swat :) |
| 10:05.28 | *** join/#asterisk xilun (~xilun@shattrath.sceen.net) |
| 10:05.30 | xilun | hi |
| 10:05.42 | RadJackson | Nop , we have tried to make calls to differents numbers from differents providers, same result it displays "Call From (+CountryCode NUMBER)" |
| 10:07.06 | *** join/#asterisk seik0 (5f349ef0@gateway/web/freenode/ip.95.52.158.240) |
| 10:09.43 | seik0 | Hi, again, everyone. Once I had a problem: SIP stops working when I loose connection to external sip provider (which is "register => ..." in sip.conf) because of external internet connection failures. Here was an advice to add DNS name of sip-providers to /etc/hosts. But that doesn't help - asterisk get stuck and only way to get it work is to unregister this sip-connections. |
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| 10:11.02 | seik0 | Somewhere in internet I meet advices to create some always working sip-provider, dummy one. So, when connection fails, then asterisk tries to find any working "register" peer and continue to work as usual |
| 10:11.13 | seik0 | But here was said, that this will not help |
| 10:11.38 | seik0 | I'm going to try. But maybe any more help on this issue? |
| 10:16.05 | sjs205 | Does asterisk have any tools for adding a sipuser to the sip database? |
| 10:16.34 | seik0 | sjs205: what is "sip database" |
| 10:17.15 | seik0 | if it's a DB table (realtime), than you need simply add new row with all required parameters |
| 10:17.16 | sjs205 | Sorry seik0, I';ve setup a realtime configuration using the mysql table structe... |
| 10:17.39 | seik0 | so, you just use any sql client and add rows |
| 10:17.41 | sjs205 | seik0, yeah, I was hoping there would be somesort of tool that would automate this process. |
| 10:17.57 | seik0 | what exactly you want automate? |
| 10:17.57 | sjs205 | like add a peer in the cli? |
| 10:18.49 | sjs205 | seik0, something like, 'sip add user alice@domain.com" |
| 10:18.53 | sjs205 | :) |
| 10:19.51 | sjs205 | Can this be done with the web management interface? |
| 10:20.20 | seik0 | I don't know of such tool. sip user had a lot of parameters to work with. but some ways may exist |
| 10:20.33 | seik0 | web interface of asterisk? |
| 10:21.22 | sjs205 | seik0, yeah, I have seen reference to some sort of portal, and now I've just found out that freePBX provides this functionality for asterisk |
| 10:22.51 | seik0 | i never used web interface for asterisk, so can't say anything |
| 10:24.51 | seik0 | i think, you can add some scripts to system to add entries to mysql DB |
| 10:25.07 | seik0 | with syntax you need |
| 10:25.26 | seik0 | the only difference will be that command run not in CLI |
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| 10:27.13 | sjs205 | seik0, yes, that is what I'm thinking of doing. It may not be so bad though, since ultimatly, my application is web based... |
| 10:27.41 | sjs205 | Do you know when I can find a complete realtime database setup docs for asterisk 1.8? |
| 10:29.11 | seik0 | start here: http://www.voip-info.org/wiki/view/Asterisk+RealTime |
| 10:29.24 | seik0 | and follow white rabbit |
| 10:29.27 | seik0 | follow links |
| 10:32.12 | sjs205 | Haha, cheers seik0... I've found those pages before but gave up because the white rabbit is quite quick at diving down those holes! |
| 10:33.00 | kaldemar | i'd definitely not start in voip-info. http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DB.html and contrib/realtime in a source package for more example tables. |
| 10:33.38 | seik0 | kaldemar, thanks |
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| 10:34.05 | sjs205 | Thanks kaldemar, yes, I have never really spent much time looking at the voip site since everything seems to outdated... |
| 10:38.31 | seik0 | By the way, I have an issue on asterisk 1.8 with Asterisk-DynamicRealtime-unixODBC--Oracle: exten-patterns (_X123X.) not working correctly, such patterns not found in DB. In asterisk 1.4 there was Ok. I resolved it by using mixed static/dynamic realtime putting in dynamic part only really dynamic extensions without such patterns |
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| 10:44.52 | seik0 | checked DB logs and got that "select ..." is quite correct, but asterisk not receiving result. Started to dig sources, but had not time unfortunately |
| 10:45.49 | seik0 | As i know, actual bestpractices says "dynamic realtime is Bad" ? =) |
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| 11:24.59 | James87 | I'm looking for some wav files to download to use as music on hold. Anyone who can advise a good website? |
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| 12:14.38 | leifmadsen | seik0: I use dynamic realtime all the time, works great when you know how to set it up :) |
| 12:15.13 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 12:17.23 | romaintordo | afternoon, I am currently running asterisk version 1.6.2 with FreePBX 2.7.0.10 for about 2 years, everything is fine, I have a new server comming in 2 weeks, do you recommend to use asterisk NOW 3.0 to replace my production machine or is it too early and I should use an older version of asteriskNOW? |
| 12:23.22 | [TK]D-Fender | romaintordo, Don't waste time going for older ones. |
| 12:23.45 | [TK]D-Fender | romaintordo, Get the latest stable release of AsteriskNOW, or the FreePBX Distro |
| 12:24.38 | romaintordo | good to hear - thanks D-Fender; I'll build my next server on 3.0 in that case. Have a good day |
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| 13:07.46 | Katty | morning |
| 13:20.22 | Captain_Proton | ? on rxgain & txgain everything I have read said to positive number so I am understanding it right that to tuen it down txgain by 8 i would put "txgain = 8" |
| 13:20.23 | Captain_Proton | or would it be txgain = -8 |
| 13:20.23 | *** join/#asterisk classix (~salven@silenceisdefeat.com) |
| 13:20.24 | igcewieling | you would use a negative number |
| 13:27.28 | Captain_Proton | igcewieling: thanks |
| 13:32.26 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 13:35.14 | Captain_Proton | I know I asked this before but to many of partying. When I change chan_dahdi.conf do I need need to do a dahdi restart or will askerisk reload do? |
| 13:39.44 | [TK]D-Fender | reload chan_dahdi.so |
| 13:39.44 | [TK]D-Fender | individually or with * as a whole. |
| 13:39.46 | Captain_Proton | will that restart the pri channels? |
| 13:39.47 | igcewieling | Captain_Proton: no. |
| 13:39.47 | igcewieling | Captain_Proton: MOST changes to chan_dahdi.conf will be applied on a reload or reload chan_dahdi.so. There are a few things which require you restart asterisk or unload / load chan_dahdi, but those items are listed when you reload the module |
| 13:39.48 | kaldemar | Captain_Proton: asterisk reload or restart, depending what you change. |
| 13:41.02 | Captain_Proton | Thanks |
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| 13:48.10 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:54.04 | xilun | if anybody is interested to test some "sip show peers" with lots of registered peers (~3500+ ; try with 4000 or 5000) , can try: https://issues.asterisk.org/jira/secure/attachment/47013/fix_sip_show_peers_stack_overflow_asterisk_11.3.0.patch ( issue : https://issues.asterisk.org/jira/browse/ASTERISK-21466 ) |
| 13:54.05 | LieutPants | [ASTERISK-21466] [Status: Triage] [crash] "sip show peers" crashes Asterisk with ~3500 registered peers - https://issues.asterisk.org/jira/browse/ASTERISK-21466 |
| 13:54.20 | Katty | guys. i have a very serious problem. |
| 13:54.23 | xilun | without the patch it crashes with stack overflow |
| 13:54.28 | Katty | it requires a very ingenious plan. |
| 13:54.31 | Katty | i am out of coffee. |
| 13:54.45 | Katty | HOW DID THIS HAPPEN!? |
| 13:58.21 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:58.21 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:58.40 | igcewieling | Katty: I blame the squirrels. |
| 13:58.52 | Katty | shakes fist |
| 13:59.15 | Katty | more coffees will be had! |
| 14:00.35 | igcewieling | LieutPants: Several sip related deadlocks were fixed in recent versions of Astersik |
| 14:00.50 | Katty | LieutPants: i love the /nick |
| 14:00.58 | igcewieling | ah, you appear to be running the latest. |
| 14:01.17 | Katty | LieutPants: but for some reason, my brain read it as lieutenant pants. |
| 14:01.30 | igcewieling | LieutPants: do you NEED to run Asterisk 11? Have you considered reverting to a more mature version of Asterisk like 1.8.x? |
| 14:03.20 | xilun | i think you are talking to a bot... |
| 14:03.29 | xilun | test : ASTERISK-21466 |
| 14:03.30 | LieutPants | [ASTERISK-21466] [Status: Triage] [crash] "sip show peers" crashes Asterisk with ~3500 registered peers - https://issues.asterisk.org/jira/browse/ASTERISK-21466 |
| 14:03.34 | xilun | yes you are :) |
| 14:03.54 | xilun | igcewieling: the same bug exists in 1.8 btw |
| 14:03.57 | xilun | (iirc) |
| 14:04.24 | igcewieling | This makes me VERY happy our peers don't register. 8-| |
| 14:04.51 | mjordan | Lieutenant Pants is the demoted Captain Pants. |
| 14:05.00 | mjordan | I blame leifmadsen |
| 14:05.22 | jmetro | captain underpants is his sidekick? |
| 14:05.54 | mjordan | xilun: most likely. Although a 'sip show peers' with 3500 peers on the CLI is a bit silly. I'm sure this is being done by an external process and is scraping the CLI output, but still. |
| 14:06.03 | mjordan | (yes, it shouldn't crash) |
| 14:06.16 | igcewieling | mjordan: AMI has similar functions, I wonder if they also have the issue. |
| 14:08.15 | mjordan | most likely |
| 14:08.18 | mjordan | oh well :-P |
| 14:08.32 | xilun | mjordan: i agree |
| 14:09.40 | xilun | in our systems, it is done every 5 minutes by a monitoring process that i don't really know what it does with the info... |
| 14:10.27 | sjs205 | I've got a number of extensions under the "[default]" section in extensions.conf, and I have created a new user with the context "default", but when I dial one of these extensions I get a [default]... any ideas? |
| 14:10.27 | xilun | and it might be done by an admin that does not know in advance there are so many peers |
| 14:11.11 | sjs205 | ooops, but when I dial one of these extensions i grejected because extension not found in context 'default' |
| 14:14.24 | *** join/#asterisk teff (~teff@212.42.177.8) |
| 14:17.02 | igcewieling | xilun: I'm pretty sure you can get registration / unregistration / lag messages via AMI events, that might be a better way to handle this. Maybe once per hour do the sip show peers to make sure. |
| 14:17.14 | [TK]D-Fender | sjs205, PASTEBIN is your friend... |
| 14:17.15 | [TK]D-Fender | ~pb |
| 14:17.16 | 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. |
| 14:17.17 | [TK]D-Fender | ^^^ |
| 14:17.48 | [TK]D-Fender | sjs205, Show su the call at verbose 10, SIP DEBUG enabled, along with your extensions.conf and "ls -la /etc/asterisk" |
| 14:17.51 | [TK]D-Fender | us* |
| 14:19.22 | xilun | igcewieling: probably, though we don't work on that part of the code right now and i'm making other tests on other parts and have a lot of user, and to finish the bug can also affect the sip show peers equivalent via AMI (the same function is used) and even if it did not it needs to be fixed anyway |
| 14:20.17 | igcewieling | xilun: *nod* In the long term watching the events should use far far fewer resources than a sip show peers every 5 mins. |
| 14:21.23 | xilun | yes, obviously |
| 14:21.43 | xilun | although it does not take too much resources |
| 14:21.59 | xilun | even with that many users |
| 14:22.09 | xilun | but still, I agree on the principle |
| 14:23.12 | *** join/#asterisk pepesmith (~pepesmith@unaffiliated/pepesmith) |
| 14:24.25 | sjs205 | [TK]D-Fender, cheers, I'm just doing it :D |
| 14:25.44 | *** join/#asterisk vlad_starkov (~vlad_star@87.117.139.75) |
| 14:26.43 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 14:36.17 | sjs205 | here we go [TK]D-Fender, apologies for the delay, got held on a call... http://pastebin.com/fUjNcUD6 |
| 14:37.37 | [TK]D-Fender | sjs205, Looking for *2111 in general (domain swannsips.com) |
| 14:37.45 | [TK]D-Fender | sjs205, that says "general', not 'default' |
| 14:38.32 | sjs205 | [TK]D-Fender, sorry, it is default... atleast I think... I changed that to try and see if it worked since I found that in a tutorial... |
| 14:38.50 | *** join/#asterisk vlad_starkov (~vlad_star@87.117.139.75) |
| 14:38.56 | [TK]D-Fender | sjs205, that debug clearly says it's looking in 'general' |
| 14:40.17 | sjs205 | I've done a reload and now it says default because I changed those test settings back... I'll post the correct one now... |
| 14:41.19 | sjs205 | [TK]D-Fender, that is the new loghttp://pastebin.com/HVCBDtdU |
| 14:41.52 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 14:42.50 | *** join/#asterisk navaismo (~navaismo@189.191.250.167) |
| 14:42.53 | *** join/#asterisk MiserySoft (~Elende@host81-139-114-226.in-addr.btopenworld.com) |
| 14:43.11 | [TK]D-Fender | sjs205, I fail to see any exten that SHOULD match what you dialed there. |
| 14:43.28 | [TK]D-Fender | sjs205, http://pastebin.com/fUjNcUD6 <- what line do you see that should match that number? |
| 14:45.33 | sjs205 | You know, I've just realised my confusion... I'm the guy that was trying to get opensips and asterisk working together... on the same server... And now I realise that that routing was from opensips... So I guess that *1111 for voicemail was from the opensips routing table... sorry about that... |
| 14:45.47 | [TK]D-Fender | sjs205, Your config is loaded with a ton of * 1.0 standard junk. |
| 14:46.02 | sjs205 | What would I dial to access the extension defined on line 834 [TK]D-Fender ? |
| 14:46.10 | [TK]D-Fender | sjs205, And is not valid on anything in the past half a decade... |
| 14:46.42 | sjs205 | Basically, 831 - 875 are my extensions... can I safely delete everything else? |
| 14:46.51 | [TK]D-Fender | sjs205, "VMR_ONEORMORECHARSAFTERTHEPRECEEDINGUNDERSCORE" |
| 14:46.59 | sjs205 | haha... Damn ubuntu standard install! |
| 14:47.05 | [TK]D-Fender | sjs205, remove the garbage. |
| 14:47.07 | AkkerKid | heya all! |
| 14:47.22 | [TK]D-Fender | sjs205, those are SAMPLE configs. You used/left them there. |
| 14:47.48 | sjs205 | [TK]D-Fender, ehh?: VMR_ONEORMORECHARSAFTERTHEPRECEEDINGUNDERSCORE |
| 14:48.11 | [TK]D-Fender | sjs205, VMR_ + One or more chars. |
| 14:48.23 | sjs205 | I will remove everything except 831 - 873 |
| 14:48.25 | sjs205 | :) |
| 14:48.36 | [TK]D-Fender | sjs205, VMR_1 , VMR_2, VMR_FRED |
| 14:48.50 | sjs205 | oh, VMR_alice should do it! :) |
| 14:49.26 | sjs205 | Once again [TK]D-Fender, you help has been invaluable... much appreciated :) |
| 14:49.58 | phix | [TK]D-Fender is the bestest |
| 14:50.09 | sjs205 | phix, agreed! |
| 14:50.36 | phix | he rang me up to help me test my echo problem, best resource to asterisk ever |
| 14:50.40 | AkkerKid | So I create a callfile with a few "Setver:" lines and when the call gets to the extension on the second leg, I don't have those variables set anymore. Do they not follow both paths of the call? |
| 14:50.49 | AkkerKid | Setvar:* |
| 14:51.07 | sjs205 | I'd get [TK]D-Fender to ring me if I could get this server up! :))) |
| 14:53.14 | phix | heh |
| 14:53.37 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
| 14:53.40 | sjs205 | [TK]D-Fender, should I leave the [general] section in there? |
| 14:53.51 | phix | just get it working sjs205 |
| 14:54.09 | [TK]D-Fender | AkkerKid, If there is a "second leg" then that is a DIALED channel, and it follows standard variable inheritance rules |
| 14:54.26 | igcewieling | AkkerKid: See Inheriting Channel Variables on http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html |
| 14:54.34 | igcewieling | Now go back and READ THE BOOK |
| 14:54.40 | [TK]D-Fender | sjs205, Yes, it is supposed to be used for only core settings, not actual "dialplan" |
| 14:59.41 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:59.41 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 15:01.29 | sjs205 | Right... I've removed all of those, I'm able to now make a call but still not able to hear anything.... :? |
| 15:01.31 | sjs205 | http://pastebin.com/Ky4QQNW1 |
| 15:02.59 | [TK]D-Fender | check your firewalls |
| 15:07.44 | sjs205 | [TK]D-Fender, yep, all firewalls disabled and still not a sound! |
| 15:08.26 | [TK]D-Fender | show us, and go validate your sound works at all. |
| 15:11.45 | *** join/#asterisk hilacha (~joel@201.82.26.110) |
| 15:11.49 | sjs205 | [TK]D-Fender, My sound on my desktop works since I hear the deafly twinkle daaaaaaaaaaaaaaaah noise when the call is initilised... http://pastebin.com/FSEiPmu2 |
| 15:15.06 | hilacha | someone knows why the te110p are so cheap (at least in brazil) and the board in digium site is so expensive? |
| 15:15.35 | Qwell | hilacha: The card you are looking at, that is cheap, is probably a crap Chinese clone. |
| 15:15.36 | hilacha | are the local boards fake? |
| 15:16.00 | [TK]D-Fender | sjs205, I'm not seeing firewall dumps from both sides... |
| 15:16.04 | hilacha | the chineses ones are identical (at least visual)? |
| 15:16.12 | Qwell | hilacha: I assure you, they are not. |
| 15:16.22 | [TK]D-Fender | hilacha, Many of the chinese knock-offs look almost the same. |
| 15:17.02 | sjs205 | [TK]D-Fender, firewall dumps? iptables -<something>? |
| 15:17.04 | hilacha | Qwell: do you know how to identify the original and the fake? |
| 15:17.28 | Qwell | hilacha: Easy. Real cards do not come free with purchase of a box of cereal. |
| 15:17.31 | *** join/#asterisk feeshon (~gaston@ool-45787011.dyn.optonline.net) |
| 15:17.52 | hilacha | Qwell: :) |
| 15:18.16 | Qwell | besides - the little bits of cereal get stuck inside all of the connectors |
| 15:18.25 | [TK]D-Fender | sjs205, "iptables --list" |
| 15:18.29 | hilacha | hehe |
| 15:18.30 | feeshon | I am looking to virtualize Asterisk on KVM, it's a very small asterisk server which the pysical server that it sits on is over kill. Any known issues with virtualizing asterisk? concerns? |
| 15:18.45 | Qwell | feeshon: Try it. See if it works in your environment. |
| 15:19.08 | Qwell | There is nothing about virtualization, in general, that would cause issues though (if you know what you're doing). |
| 15:19.25 | hilacha | now.. serious.. do you know how to do it? the pictures of the fake have the digium trademark.. |
| 15:19.42 | Qwell | hilacha: You purchase it from a reputable reseller. |
| 15:19.44 | sjs205 | [TK]D-Fender, http://pastebin.com/8Sa9HP8u |
| 15:19.46 | sjs205 | :D |
| 15:20.21 | hilacha | Qwell: Thank You. Good answer. |
| 15:21.09 | feeshon | I have virtualized several Operating systems on KVM and haven't ecountered major issues, but I have never virtualized a PBX and figured I would as if there were any major things I should be concerned about. This is in production. |
| 15:21.36 | navaismo | feeshon, Are you using PRi/Bri cards? |
| 15:21.56 | feeshon | no all sip |
| 15:22.41 | [TK]D-Fender | sjs205, first get rid of the Ringing() |
| 15:23.40 | igcewieling | feeshon: have you ever virtualized something that must service data in near-realtime? |
| 15:24.36 | igcewieling | nobody cares if it takes an extra 500ms to serve a web page, they will care if it takes an extra 500ms to service their audio on a phone call. |
| 15:24.47 | sjs205 | [TK]D-Fender, done... that is interesting,... that daaaaaaaaaaaaaaaaaaaaah tone is actually ringing |
| 15:24.48 | sjs205 | :) |
| 15:25.29 | feeshon | igcewieling: good point, and I don't think I have virtualized something like that |
| 15:25.48 | sjs205 | would the sound clip for that ringing be from twinkle or asterisk? |
| 15:26.58 | [TK]D-Fender | Should be Twinkle |
| 15:27.03 | igcewieling | feeshon: many people virtualize Asterisk and make it work, some fail, depends on many things, but with ALL SIP and no significant load it may work. |
| 15:27.25 | feeshon | igcewieling: Thank you |
| 15:27.52 | sjs205 | cool, still no audio from asterisk though :/ |
| 15:29.06 | [TK]D-Fender | sjs205, Ok, I'm out for now. Keep at it. |
| 15:29.10 | igcewieling | no audio is almost always a NAT / Direct Media issue. |
| 15:29.27 | sjs205 | Haha... cheers [TK]D-Fender |
| 15:29.47 | sjs205 | I have no nat... and by direct media you mean?? |
| 15:30.08 | sjs205 | does dahdi have anything to do with sound? |
| 15:30.16 | igcewieling | sjs205: "formely known as canreinvite |
| 15:30.43 | igcewieling | sjs205: yes dahdi has "something" to do with sound, if you are using dahdi. |
| 15:31.27 | *** join/#asterisk fkurkowski (~fkurkowsk@186.215.17.28) |
| 15:31.44 | sjs205 | igcewieling, I tried to use asterisk-dahdi from the ubuntu repos... but I can't see the module loaded... would it just be dahdi.so igcewieling |
| 15:31.45 | sjs205 | ? |
| 15:32.06 | igcewieling | sjs205: what PSTN interface card are you using? |
| 15:32.21 | sjs205 | igcewieling, I'm not... just sip |
| 15:32.26 | igcewieling | then you don't need dahdi |
| 15:32.38 | sjs205 | That is what I thought... :) |
| 15:32.44 | sjs205 | So that is that one sorted... |
| 15:34.20 | sjs205 | Are these two sound related at all? "ais/clm.c: Could not initialize cluster membership service?" "res_config_ldap.c: No directory URL or host found."?? |
| 15:34.58 | sjs205 | And this error should be okay, right? "chan_sip.c: No valid transports available, falling back to 'udp'." |
| 15:35.09 | sjs205 | At least it does fall back! :D |
| 15:37.46 | *** join/#asterisk cmendes0101 (~cmendes01@96.247.7.238) |
| 15:41.47 | *** join/#asterisk RyanTG (~Thunderbi@65.100.106.194) |
| 15:44.49 | learath | anyone use broadvoice? |
| 15:46.11 | [TK]D-Fender | learath, My guess is "lots of people" otherwise they'd have gone out of business years ago... |
| 15:46.33 | learath | true. but given the responses I get it appears none of them use asterisk :) |
| 15:46.46 | learath | it's officially a "supported platform" but the config example they provide is.. uhm. not good |
| 15:46.51 | learath | and their support is MS level |
| 15:47.26 | phix | https://sphotos-b.xx.fbcdn.net/hphotos-ash4/2497_365303673586604_1870371452_n.jpg |
| 15:47.46 | [TK]D-Fender | learath, Not sure what responses you're referring to are .... but if you have a problem to show us maybe we can help |
| 15:47.59 | learath | I've got everything working, except MWI |
| 15:48.03 | learath | which just fails silently |
| 15:48.28 | learath | if I try to explicitly register via mwi => I get a "subscription refused" |
| 15:49.03 | [TK]D-Fender | ~pb |
| 15:49.03 | 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. |
| 15:49.05 | [TK]D-Fender | ^^^ |
| 15:49.58 | Freeaqingme | Hi. 'set debug peer <trunk>' enables debugging to that host. However, how do I turn that off? |
| 15:50.21 | [TK]D-Fender | Freeaqingme, "set debug off" perhaps.... |
| 15:50.42 | learath | it's a single line - "[Mar 27 11:54:02] WARNING[9583] chan_sip.c: Host '206.15.148.180:5060' does not implement 'SUBSCRIBE' |
| 15:50.55 | Freeaqingme | right. I was looking for a way to disable the debugging for one specific host. But I guess I cannot [TK]D-Fender ? |
| 15:50.58 | learath | (the ip is the broadvoice sip server) |
| 15:51.00 | navaismo | ok thats the answer |
| 15:51.11 | [TK]D-Fender | learath, It's more than a single line when you enable SIP DEBUG.... |
| 15:51.20 | learath | hah will do |
| 15:51.38 | [TK]D-Fender | Freeaqingme, One, all, or none. |
| 15:51.47 | learath | ip, on or peer? |
| 15:51.50 | Freeaqingme | check. thanks |
| 15:52.14 | [TK]D-Fender | learath, Yes |
| 15:53.53 | learath | done, the only change was I got 5 of those lines instead of a single line |
| 15:56.55 | [TK]D-Fender | learath, Then you did it wrong |
| 15:57.01 | learath | hmm. possible. |
| 15:57.14 | [TK]D-Fender | learath, Absolute certainty |
| 15:57.51 | [TK]D-Fender | learath, Because a packet in & out is at least 6 lines all by itself. Then there is the fact that none are even so small |
| 15:58.03 | learath | ahh. my debug was getting turned off by the sip reload (to force it to resubscribe) |
| 15:58.31 | learath | unfotunately it's not much more. SIP/2.0 405 Method Not Allowed |
| 15:58.31 | learath | Via: SIP/2.0/UDP 192.168.88.254:5060;received=192.168.88.254;branch=z9hG4bK4c71bbe8;rport=5060 |
| 15:58.40 | learath | well. it's tons more information, but not useful |
| 15:58.46 | learath | Allow: INVITE,ACK,BYE,REGISTER,CANCEL,PRACK,OPTIONS,NOTIFY,REFER,UPDATE |
| 15:58.47 | [TK]D-Fender | learath, complete SIP comm in a pastebin..... |
| 15:58.51 | learath | that's possibly useful. ok |
| 16:02.00 | learath | mildly redacted - http://paste.lisp.org/display/136758 |
| 16:04.12 | [TK]D-Fender | learath, And on top of redaction you are only looking at half of the communication. |
| 16:04.24 | learath | did I miss the other half? Finding. |
| 16:04.26 | [TK]D-Fender | learath, You seem to be learning the term " complete in very small increments.... |
| 16:05.12 | learath | :) Really what I need to know is "who's figured out the magic incantation that broadvoice requires to make mwi work", and I'm afraid their packets are unlikely to reveal that |
| 16:05.19 | learath | lemme find the other half |
| 16:05.55 | igcewieling | learath: generally the only thing you should redact are plaintext passwords. Trust me, you are not interesting enough to call. |
| 16:06.58 | learath | I wasn't sure which were "hashed" passwords |
| 16:07.11 | learath | and I didn't bother redacting the phone number |
| 16:07.22 | learath | waiting for the next failure so I can capture the whole thing |
| 16:07.38 | sjs205 | [TK]D-Fender, igcewieling Twinkle is a pile of s*&t :) Sound was working all along! I'm an idiot too of course! |
| 16:07.56 | igcewieling | sjs205: All softphones suck, some more than others. |
| 16:08.10 | sjs205 | igcewieling, any that you'd recommend? |
| 16:08.26 | igcewieling | sjs205: I cannot recommend any softphones. |
| 16:08.42 | sjs205 | switched to linphone which seems aright so far... :) |
| 16:09.21 | learath | I've had remarkably consistant experience with softphones :) |
| 16:09.21 | sjs205 | I wonder if my openSIPS and Asterisk config would work with linphone! GRRR! |
| 16:10.25 | igcewieling | learath: all bad? 8-) |
| 16:10.45 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 16:10.55 | learath | well it depends. if you love Mr Roboto some are quite good |
| 16:13.40 | learath | hopefully http://paste.lisp.org/display/136759 is better? |
| 16:16.38 | [TK]D-Fender | learath, Not all MWI is subscription-based |
| 16:17.04 | [TK]D-Fender | learath, and ... Reliably Transmitting (NAT) to 206.15.148.180:5060: <- Broadvoice is NOT behind NAT. Fix your peer to them |
| 16:18.21 | phix | [TK]D-Fender: <3 |
| 16:19.07 | learath | fixed |
| 16:19.29 | [TK]D-Fender | learath, Go leave a VM via BV and watch for an unsolicited MESSAGE event |
| 16:19.51 | [TK]D-Fender | learath, * supports broadcast & subscribe methods or it's peers. BV may not. |
| 16:20.18 | sjs205 | I think I'm just going to leave my "AN_time" extension on automatic redial for the next 24 hours... tis like music to my ears! |
| 16:20.28 | learath | doing so |
| 16:21.20 | learath | ok, I see the notify |
| 16:22.11 | learath | so, that means I don't have the phone properly configured? |
| 16:23.14 | [TK]D-Fender | learath, Well "the phone" is another matter... |
| 16:23.29 | [TK]D-Fender | learath, Asterisk is talking to BV, not "the phone" (whatever that is). |
| 16:23.40 | learath | of course :) and I, being oh so smart, am using a Cisco 9951 which is so well supported and documented! |
| 16:23.43 | [TK]D-Fender | learath, You should start filling us in on the rest of the particulars now... |
| 16:24.20 | learath | but I do know it supports MWI, as I forced an MWI onto it via debugging, and the light lit up |
| 16:24.34 | learath | so what I've probably got misconfigured is the connection from the phone's sip connection and the broadvoice one? |
| 16:28.03 | *** join/#asterisk linocisco (~linocisco@203.81.72.87) |
| 16:28.07 | linocisco | hi all |
| 16:28.16 | Katty | howdy. |
| 16:28.30 | linocisco | i have cisco 7942G which was converted to sip |
| 16:28.46 | Katty | does it make toast, too? |
| 16:28.53 | Katty | could totally go for some toast. |
| 16:29.08 | linocisco | now registered to asterisk. but missed calls are shown with blinking. how can I clear those missed call entries? |
| 16:29.25 | learath | so, how do I figure out where that notify is going, or if it's going nowhere make it go somewhere? |
| 16:29.59 | igcewieling | Katty: my sourdough experiments have been going well. |
| 16:30.05 | Katty | igcewieling: oh? |
| 16:30.08 | Katty | igcewieling: go on. |
| 16:30.17 | [TK]D-Fender | learath, The phone isn't talking to BV. * is a B2BUA. |
| 16:30.28 | igcewieling | Katty: Have not ruined a batch in 2 weeks |
| 16:30.33 | Katty | WOOT! |
| 16:30.37 | Katty | fistbumps igcewieling |
| 16:31.06 | learath | I know. I'm asking about asterisk, I know how to make the phone talk to a mailbox, I set unsolicited_mailbox for my broadvoice context, which obviously is wrong |
| 16:31.09 | learath | I dunno why |
| 16:31.23 | linocisco | hi dear all, who is reading my msg? |
| 16:31.30 | learath | (and mailbox= for my phone, which I believe *is* working, but .. meh) |
| 16:32.16 | [TK]D-Fender | learath, "voicemail show users" |
| 16:33.00 | learath | shows the default users, with 0 mail, and the user I set in unsolicited_mailbox with 0 mail |
| 16:33.24 | igcewieling | learath: your phone will NEVER talk to broadvoice. ASTERISK is the Broadvoice client. |
| 16:34.02 | learath | igcewieling: I know that. Asterisk sits between and talks sip to both my phone and broadvoice. |
| 16:34.11 | learath | I'm not a *complete* idiot |
| 16:34.12 | [TK]D-Fender | learath, Well 0 mail = 0 notifications. And that's *'s VM boxex... nothing to do with BV clearly |
| 16:34.41 | jmetro | major ISP in my area just implemented an infinite loop in their routing |
| 16:34.46 | linocisco | hi all how to clear missed call blinking and call log to be cleared manually? |
| 16:35.01 | linocisco | it is on cisco 7942G |
| 16:35.28 | learath | I thought/hoped that by setting "unsolicited_mailbox" I could have asterisk update the MWI flag for my phone via a mailbox. |
| 16:35.47 | phix | Katty: you speak what I think, stop it |
| 16:35.54 | phix | and while you are at it bring me some toast too |
| 16:36.15 | phix | with some type of savoury spread on it |
| 16:36.28 | learath | vegamite or marmite? |
| 16:36.43 | jmetro | ....ew |
| 16:36.47 | phix | I am a vegemite fan, then again I am Australian and have grown up with it |
| 16:37.01 | phix | but I do prefer peenut butter / satay |
| 16:37.12 | phix | with chilli |
| 16:37.29 | phix | peenut butter with chopped chilly on toast <3 |
| 16:37.41 | phix | slightly melted |
| 16:38.18 | phix | but if I have to eat vegemite it goes great with sharp / matured chedar cheese |
| 16:38.41 | igcewieling | kind of like why Hawaiians like Spam? They don't know any better. 8-) |
| 16:38.42 | phix | 20 months+ |
| 16:39.02 | phix | I don't like pork products |
| 16:39.47 | *** join/#asterisk vlad_starkov (~vlad_star@87.117.139.75) |
| 16:40.01 | phix | it is a choice not forced on me like some people who's parents have brain washed them into believing it is evil |
| 16:40.36 | linocisco | i have cisco 7942G which was converted to sip |
| 16:40.40 | phix | people who force arbitary life choices onto people are wrong |
| 16:40.43 | linocisco | now registered to asterisk. but missed calls are shown with blinking. how can I clear those missed call entries? |
| 16:41.08 | navaismo | its that a deja vú? ^ |
| 16:41.25 | navaismo | the matrix has a glitch |
| 16:41.53 | Katty | delivers phix a piece of tasty garlic bread. |
| 16:41.59 | learath | interesting, event dump cache MWI was very handy |
| 16:42.04 | learath | showed me where my problem is |
| 16:52.32 | learath | ugh. found the issue I suspect - Mailbox: 5440165@SIP_Remote |
| 16:52.32 | learath | Context: SIP_Remote |
| 16:55.36 | learath | tk: thank you for helping me stop looking in the wrong places :) |
| 17:01.51 | *** join/#asterisk wtfitsme (~WTFitsME@asams.mserve.com) |
| 17:04.26 | [TK]D-Fender | learath, Baby steps... |
| 17:06.31 | learath | bada-boom |
| 17:06.38 | learath | unsolicited_mailbox=5440165 |
| 17:06.56 | learath | that had a @SIP_Remote, which was WRONG |
| 17:07.11 | learath | awesome, thanks |
| 17:08.33 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:12.31 | *** part/#asterisk wtfitsme (~WTFitsME@asams.mserve.com) |
| 17:19.13 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 17:45.13 | *** join/#asterisk Free99 (~Free99@cpe-66-108-105-10.nyc.res.rr.com) |
| 18:01.31 | hilacha | someone from brazil? |
| 18:01.53 | hilacha | that uses e1? |
| 18:03.34 | Qwell | ~ask |
| 18:03.34 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 18:03.46 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.130) |
| 18:08.02 | jmetro | can i ask a question? is qwell here? |
| 18:08.42 | Qwell | ~kick jmetro |
| 18:08.42 | infobot | ACTION kicks jmetro |
| 18:08.53 | jmetro | @.@ my shins! |
| 18:21.30 | slav3_kitten | well it's official |
| 18:21.43 | slav3_kitten | wisp network engineers are incompetent fools |
| 18:21.46 | learath | new coke sucks? |
| 18:21.58 | [TK]D-Fender | <jmetro> can i ask a question? is qwell here? <- that's TWO questions. |
| 18:22.14 | slav3_kitten | i got told today that they have no idea why i'm not getting the correct static IP |
| 18:22.14 | [TK]D-Fender | me gives jmetro a supplementary kick |
| 18:22.42 | slav3_kitten | if you're the top level network engineer, it's your job to know why shit aint working |
| 18:22.58 | slav3_kitten | not shrug an be all "i dunno" |
| 18:23.30 | Free99 | slav3_kitten, maybe that's how he got the job, if his company is a dilbert replica |
| 18:24.09 | slav3_kitten | i think it may be the case sadly |
| 18:24.42 | slav3_kitten | this new unifi pro is pretty boss however |
| 18:24.43 | jmetro | slav3_kitten: i can set you up with a static IP on my 20$ netgear router for half the cost of your ISP's static IP per month! |
| 18:24.55 | slav3_kitten | lol |
| 18:25.16 | jmetro | okay, your new external ip is 192.168... |
| 18:25.28 | slav3_kitten | don't make me pimp slap you |
| 18:25.43 | slav3_kitten | because i'll do it |
| 18:26.03 | jmetro | Fine fine i'll move you up to a class B... 127.0... |
| 18:26.29 | learath | ::1? |
| 18:26.41 | slav3_kitten | gimme your address, i'll be right over |
| 18:27.05 | slav3_kitten | well right after i find my purple suit and balling cane |
| 18:27.09 | jmetro | I can imagine people trying to tell eachother their IPv6 addresses over the phone. it will be faster to walk |
| 18:27.45 | slav3_kitten | it's been a day of successes and fails |
| 18:27.58 | learath | no no, ipv6 is perfect and flawless in every way! |
| 18:28.19 | slav3_kitten | fail, stripped out the screws in my 80 dollar dosimeter because there was poor thread engagement and possibly incorrect screw size |
| 18:28.34 | slav3_kitten | win, i got my unifi pro |
| 18:28.46 | slav3_kitten | fail, my isp is sucking a bag of dicks instead of fixing my gear |
| 18:28.56 | slav3_kitten | win, i got 60 dollars finally refunded |
| 18:29.16 | jmetro | neutral: those 60 dollars pay for the 80 dollar dosimeter? |
| 18:29.18 | slav3_kitten | if i get this unifi all setup an provisioned quickly it'll be epic win |
| 18:29.35 | slav3_kitten | nah, i'll just order some screws and a tap and fix the dosimeter myself |
| 18:29.51 | slav3_kitten | for the time being duct tape holds the thing together just fine |
| 18:30.05 | jmetro | i think its interesting that "torx security" is openable with a flathead |
| 18:30.25 | learath | only if you don't ever want to reuse it |
| 18:30.51 | slav3_kitten | jmetro, it's easier to take a screw driver an bust off the lil prong in the center then use regular torx |
| 18:30.58 | Free99 | learath, if you're careful you can do it several times |
| 18:31.12 | learath | only if they are not torqued down right |
| 18:31.18 | jmetro | I despise flathead screwdrivers though. |
| 18:31.20 | learath | to be fair, they almost never are. |
| 18:32.08 | Free99 | or if they have threadlock |
| 18:32.08 | slav3_kitten | jmetro, why do you hate flat head? |
| 18:32.23 | slav3_kitten | red loctite <3 |
| 18:32.55 | jmetro | slav3_kitten: because its an obnoxious crappy driver head that slips out constantly. I'd rather unscrew 6 stripped torx screws than 1 normal flathead |
| 18:33.12 | Free99 | flat head and philips =/= good. Robertson, w00t! |
| 18:34.02 | jmetro | Free99: square? |
| 18:34.15 | Free99 | ..yeah, aka square :) |
| 18:34.39 | slav3_kitten | i like torx or tri wing |
| 18:34.53 | slav3_kitten | or standard socket head cap screws |
| 18:35.14 | slav3_kitten | spline drives are cool however |
| 18:35.30 | Free99 | But how are those when you actually need to torque something to a pretty high level? Square doesn't pop out very easily |
| 18:35.36 | jmetro | One way screws make me sad. I always have a multi-function driver on me and like to disassemble things when i'm bored. |
| 18:35.39 | Free99 | haven't tried torx or tri wing |
| 18:35.53 | jmetro | Torx strips pretty easy under T10 but is reliable above |
| 18:35.55 | Free99 | if they're pan-head, use a vice grip :D |
| 18:36.13 | slav3_kitten | i use a dremel tool to make one way screws into flat blades :D |
| 18:36.21 | Free99 | LOL |
| 18:36.37 | slav3_kitten | if all else fails, plasma cutter |
| 18:36.47 | Free99 | whoa buddy |
| 18:37.05 | slav3_kitten | i got to pop off a high security pad lock with my plasma cutter for a friend whom lost his key |
| 18:37.23 | jmetro | A bolt cutter with long enough handles > any lock |
| 18:37.27 | slav3_kitten | fancy ass steel shackle was no match for the temps on the surface of the sun |
| 18:37.33 | Free99 | "friend" who "lost" his "key"...to the bank |
| 18:37.53 | slav3_kitten | nah he collects high security locks, and locked his foot locker with one |
| 18:38.32 | jagster` | mmm |
| 18:38.43 | jagster` | pointed isymphony at its $ipaddress instead of localhost |
| 18:38.49 | slav3_kitten | ugh "E: Sub-process /usr/bin/dpkg returned an error code (1) |
| 18:44.00 | slav3_kitten | seems mongod-10gen is being a cunt |
| 18:44.27 | jagster` | asterisk choked :< |
| 18:45.01 | slav3_kitten | awww |
| 18:45.54 | slav3_kitten | i currently have a 200 dollar AP next to me that's currently useless due to invalid instructions from ubiquiti on installing the controller software |
| 18:46.25 | Free99 | oh really? beel looking at some stuff from them actually |
| 18:46.26 | Free99 | *been |
| 18:46.39 | igcewieling | Free99: they are well known and known to work well |
| 18:46.40 | Free99 | I'm running their sr-71 card in my laptop in fact, really awesome |
| 18:46.56 | igcewieling | but I've had my fill of silly wifi stuff already so I'm outta here. |
| 18:46.59 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 18:47.15 | Free99 | kthxbye? |
| 18:47.15 | slav3_kitten | Free99, i love their gear |
| 18:47.36 | slav3_kitten | however mongo db is being a dick so until i sort that out i can't install their controller software |
| 18:47.41 | jmetro | Ubiquiti has VERY nice AP's |
| 18:47.54 | slav3_kitten | mhmm! |
| 18:48.09 | slav3_kitten | which is why i dropped more money then i should have on it |
| 18:48.14 | Free99 | better than [dd,open]-wrt? |
| 18:48.31 | slav3_kitten | by far |
| 18:48.52 | slav3_kitten | mikrotik makes good gear too |
| 18:49.00 | jmetro | we use lightweight ones which are good |
| 18:49.04 | slav3_kitten | but if you want really good access points, i suggest juniper |
| 18:49.15 | jmetro | i dont care about setup though because ohwow their performance is spectacular |
| 18:49.53 | jmetro | i have a feeling that ubiquiti is breaking FCC regulation with how strong those AP's are |
| 18:50.09 | Free99 | shoot, my card is 450mw max |
| 18:50.14 | slav3_kitten | jmetro, doubt it. |
| 18:50.23 | Free99 | for a minipci laptop card! |
| 18:50.42 | slav3_kitten | what's the eirp on it? |
| 18:51.01 | Free99 | and my router is a 54g with ddwrt which I waterproofed and put outside with a solar panel. |
| 18:51.15 | Free99 | ...to borrow other people's wifi :P |
| 18:51.20 | slav3_kitten | LOL |
| 18:52.13 | slav3_kitten | ffffff 10 more minutes until this mongodb downloads |
| 18:52.29 | jmetro | sounds like a perfect opportunity to play dwarf fortress |
| 18:52.52 | slav3_kitten | sounds like a perfect chance to grab a slice |
| 18:53.54 | slav3_kitten | i kind of wish i could have justified a juniper based wlan solution. i keep drooling over their features |
| 18:54.28 | *** join/#asterisk derjanni (~derjanni@ip-178-202-27-28.unitymediagroup.de) |
| 18:54.31 | *** join/#asterisk acidfu (~nib@unaffiliated/acidmen) |
| 18:54.33 | derjanni | Good evening everybody. |
| 18:54.48 | slav3_kitten | but no way i coould justify 1500 for 2 APs and a controller, for my home network |
| 18:55.19 | derjanni | I keep getting "chan_sip.c:13189 sip_reg_timeout" with my provider localphone.com, although FW is set up to fwd 5060 and 10000-20000 udp |
| 18:55.20 | derjanni | any ideas? |
| 18:55.38 | derjanni | tried nat=yes and nat=no ... no effect |
| 18:59.16 | [TK]D-Fender | derjanni, PASTEBIN your actual registration attempts with SIP DEBUG enabled. |
| 18:59.17 | [TK]D-Fender | ~pb |
| 18:59.17 | 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. |
| 18:59.19 | [TK]D-Fender | ^^^^ |
| 18:59.35 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 19:01.28 | paulc | I put a Ubiquiti UniFi in at home and love it :) |
| 19:02.05 | paulc | Next step: Replace the shitty ActionTec telco-supplied router, possibly with Mikrotik.. that's been on the to-do list for months though.. |
| 19:02.17 | derjanni | there you go: http://susepaste.org/1417148 |
| 19:02.34 | Free99 | paulc, you mean craptiontec? |
| 19:03.50 | Free99 | my dad paid for 40mbps fios, tests showed his computer only getting ~28mbps. Replaced the actiontec with a new one, and hey presto! 40mbps...for 2 weeks. then back to 28 |
| 19:04.41 | paulc | Free99: Yeah, they don't exactly have a good rep do they.... So they just replaced it with another (same) unit? |
| 19:05.12 | Free99 | yep. I mean it hooks directly to the fiber line, not like I can replace it with something else |
| 19:06.23 | tacpilot | *98 gets me to the mailbox welcome center.. is there a way to go straight to a user mailbox and by pass that ? |
| 19:06.26 | paulc | oh really? My fiber goes to an ONT which then goes ethernet to the WAN port on the ActionTec (V1000 I think?) |
| 19:06.43 | paulc | tacpilot: core show application voicemailmain |
| 19:06.58 | paulc | You can pass in mailbox number and optionally skip password prompt |
| 19:08.10 | tacpilot | dont mind password prompt .. just want a user to go straight to their box then enter their password |
| 19:12.49 | Free99 | paulc: the ONT was motorola for a while but I actually see them by actiontec now |
| 19:13.33 | Free99 | wait a sec that doesn't make sense |
| 19:13.38 | [TK]D-Fender | derjanni, Retransmitting #1 (NAT) to 94.75.247.45:5060: Contact: <sip:4314982@192.168.1.10:5060> |
| 19:13.56 | [TK]D-Fender | derjanni, Your provider is NOT behind NAT, and you have not set yours up properly for the fact that it itself is. |
| 19:14.38 | [TK]D-Fender | derjanni, "localnet=", "nat=yes", "directmedia=no", "externaddr=" all need to be set under [general], and "nat=no" in your provider's peer |
| 19:15.30 | paulc | Free99: So you don't have fiber --> ONT --> router... yours goes Fiber --> Router? |
| 19:15.31 | [TK]D-Fender | paulc, I just got a Microtik RB2011UAS-2HnD yesterday and am in the process of setting it up |
| 19:15.42 | [TK]D-Fender | paulc, Amazing value routers... |
| 19:15.51 | paulc | Free99: I know they like the ACtionTec here cos it can do ADSL or fiber+ONT |
| 19:16.13 | learath | hah - I got the RB2011UAS-RM |
| 19:16.25 | paulc | [TK]D-Fender yeah - I've got an RB750GL at my desk here at work, been playing with it purely for DHCP/Option 66 for provisioning, on a network that already has that in place.. "handy" :-) |
| 19:16.33 | [TK]D-Fender | learath, I'm considering one of those for my office |
| 19:16.38 | Free99 | paulc, nah, I must be wrong. the actiontec is integrated into the wallmount ont somehow, but the ont was seperate, though physically connected |
| 19:16.46 | learath | if you get one check the SFP port |
| 19:16.53 | learath | the first one I got had internal damage |
| 19:17.05 | *** join/#asterisk Foxi352 (~Foxi352@v-172-252.access.restena.lu) |
| 19:17.10 | [TK]D-Fender | learath, But perhaps more for our secondary link (FTP / alt) as it lacks certain modularity for IDS / content filtering |
| 19:17.34 | learath | I'm using it as a home router, so it's perfect |
| 19:18.05 | Free99 | poll: you're providing sailors at sea with their only source of internet. Do you block porn? |
| 19:18.12 | paulc | Free99: Like physically connected with ethernet? or more like snap together modules? (I think our ONTs are Alcatel/Lucent) |
| 19:18.20 | learath | seems more effort than it's worth |
| 19:18.21 | paulc | Free99: No, but maybe rate limit it? ;-) |
| 19:18.29 | Free99 | paulc, snap together |
| 19:18.33 | [TK]D-Fender | learath, Absolutely. If I had a rack I'd have thought of it as well. The pricepoint for the RM is insane... |
| 19:18.57 | paulc | We had a request from our offshore call center yesterday to block web browsing from agent PCs.. collective eyes rolled - we're not going to solve a human issue with technology - manage your staff already! |
| 19:18.57 | Free99 | right? that's what I said. boss is definitely religious to where personal views are being put on others :-/ |
| 19:19.39 | Free99 | i mean these are industrial sailors, when do they bring women with them? |
| 19:19.43 | Free99 | lol |
| 19:21.59 | paulc | what kind of bandwidth is there on the ship? |
| 19:23.28 | Free99 | iunno, like 128kbps. VoIP will only work if I force g729 |
| 19:23.40 | Free99 | varies with weather too |
| 19:23.48 | Free99 | but I figure QOS can handle that |
| 19:25.13 | paulc | Yeah.. got your work cut out for you there eh :) |
| 19:28.06 | Free99 | guess so |
| 19:28.42 | Free99 | k, just wanted to see if anyone else agreed. I can now go to boss and say "guys and maybe girls, i don't know b/c its internet, back me up on not blocking porn" |
| 19:28.48 | slav3_kitten | this unifi shit is awesome :D |
| 19:29.53 | paulc | Free99: Yeah.. blocking ends up being a cat'n'mouse game ultimately |
| 19:30.04 | paulc | slav3_kitten: what you got? |
| 19:30.39 | slav3_kitten | unifi pro access point |
| 19:31.14 | slav3_kitten | i /should/ be configuring it for quick tossing it in an calling it good instead i'm playing with all the options as it hangs out on my desk |
| 19:33.34 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 19:33.34 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 19:33.53 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-rlhqrdrufgkzizkh) |
| 19:33.53 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 19:33.53 | *** join/#asterisk ipengineer (~ipenginee@static-71-252-134-63.dllstx.fios.verizon.net) |
| 19:34.11 | slav3_kitten | you know how it is with new toys |
| 19:34.22 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 19:35.07 | ipengineer | Has anyone come across this error before? No such switch 'Realtime' |
| 19:35.45 | ipengineer | I noticed I do not have a module pbx_realtime.so I am not sure if that is still needed in 11.3 |
| 19:37.31 | *** join/#asterisk classix (~salven@silenceisdefeat.com) |
| 19:38.14 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 19:39.36 | Qwell | ipengineer: Are you trying to use a realtime switch? |
| 19:39.49 | ipengineer | Qwell: Yes |
| 19:40.06 | Qwell | then it would help to have the module that implements realtime switches, yes |
| 19:40.50 | ipengineer | Qwell: so the two modules I have are res_realtime and func_realtime.. There are others I am assuming? |
| 19:50.05 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 19:52.22 | danfromuk | Using realtime extensions, is there any way to get around the requirement of having the switch=>realtime/@ lines? |
| 19:54.29 | *** join/#asterisk classix (~salven@silenceisdefeat.com) |
| 19:57.15 | derjanni | [TK]D-Fender added localnet, net, directmedia, externaddr - still no luck :-\ |
| 19:58.34 | derjanni | in iftop I see outbound data, but no inbound from localphone.com |
| 19:59.05 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 19:59.25 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 20:03.19 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 20:03.39 | [TK]D-Fender | derjanni, You've made changes apparently, but never showed me the new configs or the new call. Please correct this... |
| 20:04.00 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 20:06.22 | *** join/#asterisk Pumuki (~fermat@62.97.71.6) |
| 20:07.31 | derjanni | [TK]D-Fender: here is my config: http://susepaste.org/38570933 |
| 20:09.13 | derjanni | and here is my sip log: http://susepaste.org/55688680 |
| 20:13.55 | *** part/#asterisk tacpilot (~tcope@c-76-31-175-168.hsd1.tx.comcast.net) |
| 20:17.44 | Pumuki | hi! After a few days of looking up about how to config a pri card's (te122) B channels, I was wondering if you might know by heart: what's the way to make a few of those channels work for voice and others for data? (ie: 12 voice + 12 data, or 22 voice + 2 data) - I don't mean signaling channel (D) but B ones :) |
| 20:18.00 | Pumuki | thanks in advance :) |
| 20:22.10 | [TK]D-Fender | derjanni, make sure aal SIP helpers/proxies, etc are disabled on your routers. Also ensure that 5060,10000-20000 all UDP are forwarded to your server |
| 20:22.49 | [TK]D-Fender | Pumuki, voip-info Wiki has some info on that. It's almost never used... |
| 20:23.22 | [TK]D-Fender | Pumuki, http://www.voip-info.org/wiki/view/Asterisk+Data+Configuration |
| 20:23.39 | [TK]D-Fender | Pumuki, This would have to be updated slightly for more modern syntax, etc |
| 20:23.41 | derjanni | [TK]D-Fender no SIP helpers/ proxies - the router is a nude OpenWRT WR1043ND |
| 20:23.52 | derjanni | no gui, nothing |
| 20:24.02 | [TK]D-Fender | derjanni, OpenWTR is often an offender all by itself |
| 20:24.35 | derjanni | is there a chance to use TCP / STUN instead of UDP? |
| 20:25.01 | [TK]D-Fender | derjanni, Go test without the WRT |
| 20:25.03 | [TK]D-Fender | checkout time, BBIAB |
| 20:26.00 | derjanni | Can't test it without the WRT, its on a Raspberry Pi box and required DOCSIS network access |
| 20:28.34 | jmetro | so Lync is just a hosted cisco call manager |
| 20:36.58 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 20:36.59 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 20:40.42 | jmetro | Anyone know if the Cx600 are Asterisk capable? i cant imagine why they wouldnt be |
| 20:40.46 | jmetro | polycom |
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| 21:16.42 | fireman_biff | When I try to load chan_dahdi.so in asterisk it fails, with a log entry of "loader.c: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: pri_persistent_layer2_option". Any ideas why? |
| 21:17.11 | WIMPy | Wrong libpri perhaps? |
| 21:17.21 | WIMPy | How did you install the stuff? |
| 21:18.31 | fireman_biff | I didn't... somebody else installed elastix, and I believe they upgraded dahdi |
| 21:18.50 | WIMPy | ohoh |
| 21:19.19 | fireman_biff | other thing is, it doesn't even have a pri |
| 21:19.21 | fireman_biff | just analog |
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| 21:19.52 | WIMPy | But that dahdi mudule obviouselt has "pri" support. |
| 21:24.37 | fireman_biff | oh so there's nothing pri-related i can disable as a work around then... do you think the version of asterisk could have anything to do with it? or would it just be on the dahdi/libpri side? |
| 21:24.47 | igcewieling | fireman_biff: welcome to Package Hell. |
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| 21:25.06 | fireman_biff | cause we have another pbx with the same version of dahdi/libpri, but a different version of asterisk, and that one is working |
| 21:25.21 | igcewieling | fireman_biff: you have a version mismatch between the version of libpri asterisk was built for and the version of libpri installed in your system |
| 21:25.34 | jmetro | Has anyone ever had the Directory application refuse to do First & Last names for searching? Mine wont do anything besides last name only. |
| 21:25.45 | igcewieling | jmetro: yes. |
| 21:26.23 | jmetro | igcewieling: Whats the issue happening? |
| 21:26.27 | igcewieling | For example a name of "John Doe Jr" the "last name" is "Jr" |
| 21:27.05 | jmetro | I have a "John Doe" that cant search for anything besides "Doe" |
| 21:27.45 | igcewieling | jmetro: that issue I've not seen. what are the options Directory is called with |
| 21:28.07 | fireman_biff | WIMPy and igcewieling: upgrading libpri fixed the issue, thanks for pointing us in the right direction |
| 21:28.17 | jmetro | Directory(context,b) like it should be |
| 21:29.26 | igcewieling | jmetro: on MY asterisk: Directory([vm-context][,dial-context[,options]]) |
| 21:29.37 | jmetro | yeah i had to add a second comma |
| 21:29.37 | igcewieling | which means your "dial context" is "b" |
| 21:30.01 | jmetro | I must have gotten used to coding in a language that wasnt so literal =p |
| 21:30.46 | igcewieling | jmetro: does "f" or "l" options work? |
| 21:31.05 | jmetro | i did Context,,b and it worked on the first try |
| 21:32.48 | igcewieling | so it is working now? |
| 21:32.55 | jmetro | Yep |
| 21:33.26 | jmetro | your question about the parameters made me double-take and wonder if asterisk was literally making my dial-context "b" which it was. |
| 21:33.47 | igcewieling | Yay! Work day is over, I can actually accomplish something now |
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| 21:38.50 | jagster` | i volunteered to install our zimbra certs just so i could feel like i accomplished something |
| 21:38.54 | jagster` | cus fuuuu asterisk reporting |
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| 21:52.37 | igcewieling | jagster`: we wrote our own reporting scripts |
| 21:58.12 | jagster` | so did "we" thats where the pain comes from :P |
| 21:59.11 | igcewieling | heh. Our reporting was pretty easy to do. We generate CEL events in our dialplan, then use that data for our reports |
| 21:59.31 | jmetro | igcewieling: is there a way to limit an extension from being listed in the directory? |
| 21:59.54 | *** join/#asterisk forgotmynick (5006f7c1@gateway/web/freenode/ip.80.6.247.193) |
| 21:59.56 | igcewieling | jmetro: see voicemail.conf.sample if there is a way, it would be listed in there. |
| 22:00.00 | forgotmynick | hello |
| 22:00.28 | WIMPy | Nice nick |
| 22:00.31 | forgotmynick | i'm having trouble working with getonsip.com. I've added a trunk - the trunk is online but it's not showing as registered |
| 22:00.34 | forgotmynick | thanks ^^ |
| 22:00.51 | forgotmynick | is the auth string supposed to be username:pass:authusername@blah.com? |
| 22:01.38 | WIMPy | register => [transport://]user[:secret[:authuser]]@domain[:port][/extension][~expiry] |
| 22:01.57 | jagster` | being as how old this is theyre using dcontext as the pivot for call transfers |
| 22:01.57 | WIMPy | As listed in the sample config. |
| 22:02.03 | jagster` | and no call event logging :( |
| 22:02.05 | igcewieling | Golly WIMPy where did you get THAT infor from? |
| 22:02.28 | WIMPy | Magic |
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| 22:03.10 | igcewieling | jagster`: give up. give up now. |
| 22:03.21 | jagster` | igcewieling: :> |
| 22:03.41 | jagster` | ive got enough steam to upgrade but i dont know the system enough yet to upgrade safely |
| 22:03.41 | igcewieling | where "give up" could mean "upgrade to 1.8" |
| 22:03.57 | igcewieling | jagster`: FreePBX or some other GUI? |
| 22:04.11 | WIMPy | old stuff |
| 22:04.36 | jagster` | worse, older |
| 22:04.38 | jagster` | trixbox |
| 22:04.44 | jagster` | with of course freepbx |
| 22:05.03 | igcewieling | Wow, your life really sucks, doesn't it? |
| 22:05.14 | jagster` | $bigboss is asking for progress |
| 22:05.16 | jagster` | -_-'' |
| 22:05.32 | jagster` | "welp ive laid out the ground work for an upgrade but uh yeah" |
| 22:05.54 | igcewieling | jagster`: your problem is one of the major problems recent asterisk versions try to resolve |
| 22:06.13 | jagster` | yep im told "there were some missing features we require" |
| 22:06.18 | jagster` | however no one knows what they were |
| 22:06.27 | jagster` | so i cant just go look up a features list lol |
| 22:06.30 | igcewieling | linkedid is one of them. |
| 22:06.35 | igcewieling | cel is another |
| 22:06.36 | jagster` | on newer versions of trixbox |
| 22:06.56 | jagster` | yeah they said they kept an older version because newer versions of asterisk did not have what they needed |
| 22:06.57 | igcewieling | you would want to go with plain FreePBX if you must use a GUI |
| 22:07.08 | jagster` | the only one i got concretely was rrmemory which i come to find out just got renamed round robin |
| 22:07.25 | jagster` | everything else was "uh i dont remember" |
| 22:07.34 | igcewieling | jagster`: it is less the features of the GUI and more of the basic asterisk stuff |
| 22:07.37 | jagster` | so upgrading wont be easy and probably not painless |
| 22:07.48 | igcewieling | don't upgrade. Install a new server |
| 22:07.51 | jagster` | since i dont have anything scoped |
| 22:08.04 | jagster` | and i dont know dial plans yet to be able to recreate them on the fly |
| 22:08.15 | jagster` | ie no new server for a month or two |
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| 22:08.26 | WIMPy | Yes, a replacement seems to be less painfull. |
| 22:08.41 | jagster` | yep either that or im completely rewriting our reporting scheme |
| 22:08.51 | jagster` | also suspecting this isymphony plugin is causing other abnormalities |
| 22:09.09 | jagster` | was easier supporting an inhouse appw hen i could send it to the devs :P |
| 22:09.13 | jagster` | u wrote this pos fix it! |
| 22:09.16 | igcewieling | jagster`: then underlying asterisk system has no way to associate the two legs of a transferred call and nothing in a GUI will fix that |
| 22:09.45 | jagster` | igcewieling: whats the version of asterisk that first addressed this issue? |
| 22:10.08 | jagster` | ie what is the oldest version i can use |
| 22:10.36 | igcewieling | jagster`: I leave that as an exercise for the reader. Reading all the UPGRADE-*.txt files (they are not that long) will give you an idea of what major changes happened between versions |
| 22:10.56 | igcewieling | 1.8 has cel and linkedit, I don't know about 1.6 |
| 22:11.02 | WIMPy | Why do yu want an old version? |
| 22:12.32 | jagster` | WIMPy: only ask about the first version to introduce the fix so i can compare the different versions |
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| 22:33.13 | Rahail | Hi can some help to create global count dialplan where i can loadblance between few trunk/peer |
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| 22:45.00 | lvlolvlo | hello! |
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| 22:46.16 | Rahail | hi |
| 22:46.22 | lvlolvlo | anyone here know if it is possible to modify a *.conf file in Asterisk 11.3 for the "T38MaxBitRate" and "T38FaxMaxDatagram" or is this hard coded into Asterisk |
| 22:48.40 | lvlolvlo | hi Rahail |
| 22:49.58 | Rahail | how are you |
| 22:50.11 | Rahail | sorry I have no idea about your conf i am new to for htis |
| 22:50.31 | Rahail | last few days i been trying to do loadbalance between 2 sip trunk |
| 23:00.33 | navaismo | lvlolvlo, in the dialplan you can configure the FAXOPT(maxrate || minrate) not sure if the same for the T38MaxbitRate |
| 23:01.15 | igcewieling | "core show function FAXOPT" would tell you |
| 23:01.42 | Rahail | igcewieling can you help me please |
| 23:01.47 | igcewieling | lvlolvlo: did you read the UPGRADE-*.txt files, major changes would be listed there. |
| 23:01.48 | Rahail | i still couldnt figerout out |
| 23:01.56 | igcewieling | Rahail: you are far, far beyond my help. |
| 23:02.18 | navaismo | Rahail, depend on your need you can use the AstDB or other kind of DB to balance |
| 23:03.07 | Rahail | I want do it in plain asterisk so I do not put to much resource in that small box |
| 23:04.40 | Rahail | igcewieling maby you can just give me small live example |
| 23:04.48 | Rahail | so i can try pratice on that |
| 23:04.52 | Rahail | if you have time igcewieling |
| 23:05.52 | igcewieling | Rahail: http://www.voip-info.org/wiki/view/Asterisk+func+group |
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| 23:10.39 | lvlolvlo | thanks navaismo and igcewieling |
| 23:10.46 | lvlolvlo | sorry got d/c there |
| 23:10.51 | lvlolvlo | i'll take a look |
| 23:11.16 | navaismo | then AstDb is part of asterisk |
| 23:11.49 | Rahail | igcewieling thank you |
| 23:11.52 | Rahail | i hope i can figerout |
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| 23:30.45 | forgotmynick | thanks everyone for a few hours ago. it was the provider blocking registrations from asterisk, i found an alternative provider |
| 23:31.57 | Rahail | igcewieling can i use queue |
| 23:32.03 | Rahail | on this global thing not getting in my head |
| 23:32.11 | Rahail | http://www.voip-info.org/wiki/view/Asterisk+call+queues |
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