| 00:14.18 | dwayne | Auckla, svn info http://svn.asterisk.org/svn/asterisk/tags/1.4.39.1 |
| 00:14.53 | dwayne | Jan 17, 2011 (last change) |
| 00:15.32 | WIMPy | Or maybe https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 00:45.00 | Rahoul | hi - I am trying to figure out a dial plan where I can call from an ext into VoiceMailMain and not be asked for the mailbox? I am trying to figure out a way to pass mailbox number into Voicemailmain() any suggestions? |
| 00:45.48 | jpsharp | Pass it as an option to VoiceMailMain |
| 00:46.39 | Rahoul | I am trying to make it automatic, so that every extension can dial "abc" and the mailbox is passed automatically... |
| 00:46.52 | jpsharp | Oh. |
| 00:47.27 | jpsharp | Oh, you want to pass voicemailmail the callerid number then? |
| 00:47.34 | Rahoul | yes! |
| 00:47.43 | Rahoul | here is my setup |
| 00:47.52 | Rahoul | peers are defined in letter, for example |
| 00:48.02 | Rahoul | [front-desk] |
| 00:48.25 | Rahoul | exten => 100,1,Dial(SIP/front-desk) |
| 00:48.33 | jpsharp | exten => 4242,1,VoiceMailMain(${CALLERID(num)}) |
| 00:48.42 | Rahoul | hmmm ok... |
| 00:48.49 | Rahoul | did not think of that |
| 00:49.25 | jpsharp | or ${CALLREID(num)}@context if your mailboxes aren't in the same context as your dial commands. |
| 00:49.26 | Rahoul | let me try it |
| 00:49.34 | Rahoul | yes they are... |
| 00:49.54 | Rahoul | I was trying exten => *97,n,VoiceMailMain(${EXTEN}) |
| 00:49.59 | Rahoul | but it was passing "*97" |
| 00:50.04 | Rahoul | let me try your idea |
| 00:50.26 | jpsharp | You were passing ${EXTEN} rather than ${CALLERID(num)} |
| 00:51.15 | Rahoul | ok I tried it and it is still passing the peer name, which in my case is "front-desk" |
| 00:51.56 | Rahoul | maybe i need to set the callerid under the peer options? |
| 00:52.01 | jpsharp | Yes. |
| 00:52.21 | Rahoul | CALLERID=123? |
| 00:52.27 | Rahoul | or CID= |
| 00:53.24 | jpsharp | In the sip.conf peer, call it "callerid = Front Desk <123-345-4242>" |
| 00:53.46 | jpsharp | Then when your front desk calls in, it'll go to voicemailmain with 1233454242 for the mailbox number. |
| 00:54.03 | Rahoul | it worked like charm |
| 00:54.23 | Rahoul | thanks |
| 00:54.53 | Rahoul | Is there a way to get a listing (and ideally detail on) each of these variables? |
| 00:55.10 | Rahoul | (like ${CALLERID(num)] |
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| 01:40.09 | jpsharp | Rahoul: http://www.voip-info.org/wiki/view/Asterisk+variables |
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| 01:41.58 | shawny_ | Good evening all! Can anyone give me a quick idea of how I can do a PLAR line in asterisk using SIP. No FXS/FXO....just SIP. |
| 01:43.36 | jpsharp | YOu'll need a phone smart enough to dial a number as soon as the handset is lifted. |
| 01:44.02 | shawny_ | argh.... |
| 01:45.47 | shawny_ | is there anything that can be done in asterisk (I don't have any telephony cards for it) that will let me do a PLAR? |
| 01:46.52 | [TK]D-Fender | PLAR? |
| 01:47.22 | shawny_ | bat-phone, hotline, ring-down |
| 01:48.25 | shawny_ | ....Private Line Automatic Ringdown (can't remember if that is the exact acronym) |
| 01:50.44 | shawny_ | or, if it is easy, I need that sort of functionality mapped to a button. kind of like a speed-dial that would dial a certain number, but could also recieve a call on that button as well |
| 01:58.58 | [TK]D-Fender | shawny_: ASTERISK doesn't do that... the phone does. |
| 01:59.19 | [TK]D-Fender | shawny_: And tons of phones support that feature... Linksys & Polycom both do, plenty of others I'm sure |
| 02:01.10 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 02:04.25 | [TK]D-Fender | shawny_: As do all of the Linksys ATA's |
| 02:18.04 | shawny_ | hmm, alright, can anyone suggest an Android SIP softphone with this capability? (I know I'm being needy, but I'm trying to test some stuff, and was hoping to spend more time on the testing) |
| 02:23.59 | Rahoul | try sipdroid it's worked well for me :) |
| 02:26.55 | shawny_ | yeah, I have sipdroid and just tried 3CX. so what I need to be able to do is set the client up in such a way that it would dial as soon as I go 'off-hook' sort of like prepending a 9 from the phone itself when calling off the system? |
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| 02:48.00 | hesco1 | I'm guessing that it makes sense to install multiple formats of the sound prompts in order to save the load on the server that would be spent transforming the audio formats to for bridging with inbound calls. Am I off on that? Is there any performance benefit in that way to installing multiple formats of an audio prompt? |
| 02:53.51 | jpsharp | You get the benefit of not having to transcode, you're right. |
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| 02:59.59 | leifmadsen | what jpsharp said |
| 03:00.27 | rroche | Hi I'm having an issue trying to call ami originate here is a link of my command https://gist.github.com/3831217, I need some help |
| 03:02.38 | ChannelZ | What's the issue? |
| 03:02.43 | leifmadsen | rroche: https://github.com/russellb/amiutils |
| 03:03.21 | rroche | thanks @leifmadsen been looking for a python lib |
| 03:03.33 | ChannelZ | Well actually it looks like you're trying to dial SIP/phonenumber which has no actual meaning |
| 03:04.10 | ChannelZ | It should be SIP/peer/number or similar, unless you have a peer called "016642317252" and were intending to just dial it. |
| 03:05.24 | RiceCracker | just to confirm if i have a yellow alarm on my dahdi card, that can mean the IDAP connection with my dahdi card is expericing issues. external incoming and outgoing calls cannot work, internal calls are not a problem |
| 03:05.26 | rroche | so what would be the right way of calling out? SIP/100/phonumber ? |
| 03:05.43 | RiceCracker | ? |
| 03:18.18 | jpsharp | yellow alarm says you've got T1 issues somewhere. |
| 03:26.17 | rroche | @ChannelZ so how would you go about calling out using the Originate command |
| 03:27.41 | rroche | SIP/trunkname/phonenumber ? |
| 03:27.47 | hesco1 | thanks jsharpe, leifmadsen. I'm still pushing through this install. I have installed both mysql client and odbc but at the menu select step I still do not see access to the mysql bindings. What am I missing here. |
| 03:30.00 | jpsharp | MySQL stuff is only in asterisk-addons. |
| 03:31.27 | hesco1 | ah, ok, back to the cli, then. |
| 03:34.28 | hesco1 | I'm not seeing that here: http://www.asterisk.org/downloads where do I want to look? |
| 03:35.04 | hesco1 | except for Allison-Add-Ons, which look like voice prompts |
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| 03:55.12 | hesco1 | how long after starting asterisk would one expect to see sip become available? |
| 03:55.28 | *** join/#asterisk jmetro (~jmetro@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 03:55.38 | hesco1 | fresh source install of Asterisk 10.8.0 |
| 03:55.56 | jmetro | so is there any way to export your running dialplan with included classes |
| 03:57.29 | jmetro | i know dialplan save does extensions.conf but i -really- need to save an included "asdfextensions.conf" that i have modified but not reloaded. |
| 03:58.12 | hesco1 | I just untarred my legacy configuration into /etc/asterisk of a new install. so I am hopeful that is possible. |
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| 04:05.09 | hesco1 | I have been waiting ten minutes and sip is still not available to me |
| 04:11.14 | jpsharp | can you do a "sip show peers"? |
| 04:15.38 | ChannelZ | rroche: yes |
| 04:16.05 | ChannelZ | hesco1: what do you mean "sip is still not available to me" |
| 04:17.55 | jpsharp | He has a fresh source install of 10.8. I'm willing to bet that chan_sip.so didn't get built because of no libssl & libssl-dev |
| 04:19.48 | ChannelZ | I don't think those aren't requirements |
| 04:19.54 | ChannelZ | err are |
| 04:20.32 | *** part/#asterisk rbowles (~bb@75-147-63-201-NewEngland.hfc.comcastbusiness.net) |
| 04:24.19 | jpsharp | They are for res_crypto and res_cryto is a requirement for chan_sip now. |
| 04:24.29 | jpsharp | been bit by that problem twice now. |
| 04:24.40 | jmetro | does anyone know about the dialplan export I posted about earlier? |
| 04:25.16 | jmetro | Dialplan save works for the extensions.conf but I would -love- to export the <includes> ive done |
| 04:26.06 | ChannelZ | hmm.. my config says "can use res_cypto" not "depends on" |
| 04:27.14 | hesco1 | jpsharpe: I tracked it down. |
| 04:28.34 | hesco1 | found that my legacy sip.conf was a symlink to a file inside of a freepbx installation. once I had migrated that, and restarted the service, sip fired up immediately and my trunks all registered. |
| 04:29.03 | hesco1 | I'm now looking up the default password for a polycom so I can log in and point it at the new ip address. |
| 04:30.57 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 04:31.14 | hesco1 | Polycom : 456 |
| 04:31.23 | hesco1 | had forgotten to capitalize Polycom |
| 04:32.00 | ChannelZ | jmetro: I've never used dialplan save - ever - but includes are just like C preprocessor includes, they aren't "a thing" - The dialplan gets loaded up as one giant unit, it doesn't keep track that any particular extensions/contexts came from a particular #include |
| 04:32.21 | jpsharp | hesco1: Do you still have that test account pointed at asterisk.fivecats.org? Can you delete that? |
| 04:32.40 | jmetro | Alright.. I used the dialplan save and didnt notice if it had just appended the includes to the bottom of the file. |
| 04:33.58 | ChannelZ | mine just says "I can't save dialplan now". *shrug* |
| 04:40.16 | jmetro | It did actually append everything to the bottom of my dialplan. At least it gives me something to go off. |
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| 04:47.07 | hesco1 | yes, sure, I will. |
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| 04:56.32 | hesco1 | jsharpe: done, I have deleted it from the database of the freepbx installation of the legacy server, as well as from the migrated sip_additional.conf and sip_registrations.conf files on the new server. |
| 04:57.04 | hesco1 | I will try to remember to drop that trunk from the database on the new server once I get around to restoring that as well. |
| 04:57.30 | hesco1 | thanks for your help trying to sort out that issue. |
| 05:04.07 | *** join/#asterisk l-fy (~diana@yate/developer/l-fy) |
| 05:04.09 | l-fy | hello |
| 05:05.06 | l-fy | any asterisk user in San Francisco Bay Area? |
| 05:11.56 | hesco1 | one or three, I'd bet. |
| 05:12.56 | hesco1 | jsharpe: sorry, just now ran `sip reload` on new server, so if you were testing that, please try again. But either way it should now be off. |
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| 06:40.31 | schmidts | good morning |
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| 07:11.56 | ChannelZ | feh |
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| 08:20.58 | pbxMan | hello |
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| 09:29.28 | hurdman | hi. Is there a maximum limit of simultaneous call with asterisk ? I get a expodential top and my asterisk seems to crash aleatoiry when i make about 200 calls in the same time. |
| 09:31.34 | wdoekes | shouldn't be.. I've heard of people having lots more calls. but since asterisk is very configurable, all sorts of things could bog it down in your particular setup |
| 09:33.10 | hurdman | arf, thx. Using a lot of call files is very expensive for asterisk ? |
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| 09:34.23 | Dovid | hi all |
| 09:37.33 | wdoekes | it's not the most efficient way of setting up calls.. but I don't think it should be that expensive |
| 09:38.50 | hurdman | wdoekes: ok |
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| 10:13.29 | schmidts | call files could be expensive but it depends mostly what you are doing with these call files. do you only start two call legs which then are bridged together or do you use conferences ... |
| 10:13.34 | schmidts | btw hurdman which version? |
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| 10:37.35 | pea | hi |
| 10:39.25 | pea | i have a pb when i try to launch asterisk -r as the asterisk user |
| 10:40.03 | bulkorok | asterisk .r is not starting asterisk... |
| 10:40.05 | bulkorok | -r |
| 10:40.16 | pea | it said me: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 10:40.21 | pea | bulkorok: yep i know |
| 10:40.27 | pea | asterisk is already running |
| 10:40.29 | *** join/#asterisk Smirker (~x@124-171-218-5.dyn.iinet.net.au) |
| 10:40.36 | schmidts | and does it exists and is the process owend by the asterisk user? |
| 10:40.46 | schmidts | *owned |
| 10:41.01 | pea | the file exists and perms/owner seems to be good |
| 10:41.04 | pea | http://pastebin.com/AihWTjnu |
| 10:41.19 | pea | with the root user, asterisk -r works fine |
| 10:42.02 | pea | atm only root can acces the socket |
| 10:42.12 | kaldemar | you're running asterisk as root then. |
| 10:43.18 | bulkorok | try starting asterisk with -U and -G option... |
| 10:43.24 | pea | http://pastebin.com/FapTJ35q |
| 10:43.29 | Smirker | hi all. question about phone networks in general, specifically about caller ids. in Australia, i have a 1300 service (virtual number) pointed to a DID i have on my Asterisk box through my SIP provider. the strange thing is, 99% of the 'callerids' in the 1300 CDR and the Asterisk CDR match (based on date/time/duration of the call), however there are A FEW pairs of CDRs for the same calls that |
| 10:43.30 | Smirker | have slightly different caller ids. |
| 10:43.43 | pea | asterisk is started with /etc/init.d/asterisk start (on FreePBX) |
| 10:43.52 | Smirker | e.g. 1300 caller id: 0298530600; Asterisk caller id: 0298530622 |
| 10:44.04 | Smirker | is it possible for one call to somehow have two caller ids? |
| 10:44.07 | pea | kaldemar: http://pastebin.com/FapTJ35q |
| 10:44.22 | Smirker | (these calls are from telemarketing companies, btw, with the strange caller id behaviour) |
| 10:44.48 | kaldemar | pea: so it runs as root. |
| 10:45.02 | bulkorok | Smirker: CLIP no screening can do sth like that... |
| 10:45.31 | bulkorok | pea: check if you can give the start-script user and gruop |
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| 10:47.37 | bulkorok | pea: in my setup it's the /etc/defaults/asterisk file where the AST_USER and AST_GROUP is set and taken by the start-script |
| 10:48.04 | pea | kaldemar, bulkorok : right, it seems the startup script have a pb.. it must check /etc/sysconfig/asterisk which contains AST_USER and AST_GROUP |
| 10:48.26 | pea | weird.. |
| 10:49.34 | pea | bulkorok: are you running asterisk on a Freepbx too ? |
| 10:49.49 | bulkorok | no... debian squeeze |
| 10:49.55 | pea | ok |
| 10:51.15 | Smirker | bulkorok: interesting. i always thought a call could only have one caller id number. would you know a resource i could read that outlines specifically what information is typically carried over carrier networks (such as these multiple caller ids/etc). i wouldn't know where to start looking for something in-depth. |
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| 10:56.55 | bulkorok | Smirker: nothing english... |
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| 10:59.14 | Smirker | could one potentially be CLIP and one COLP? |
| 11:00.41 | bulkorok | sure... |
| 11:00.58 | bulkorok | COLP is at the calling side... CLIP is at the called side... |
| 11:01.17 | bulkorok | "easy" said... |
| 11:01.53 | bulkorok | and yes... COLP no screening esxists too ;-) |
| 11:03.00 | Smirker | i presume the three consequitive fullstops indicates sarcasm. which i deserve after reading what COLP actually is. |
| 11:03.36 | Smirker | ah, this is just playing with my CDR matching. guess i'll just add a second pass which ignores the caller id. |
| 11:04.26 | bulkorok | at least check the CALLERID(xx) function wich handles the callerid presentation |
| 11:04.35 | bulkorok | you can get and set whatever you need |
| 11:07.55 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 11:22.59 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 11:24.17 | tompaw | lol who changed AMI prompt from 1.1 to 1.2? :> |
| 13:01.29 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 13:01.29 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.8.0 (2012/09/13), 1.8.16.0 (2012/09/13), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 13:03.15 | *** join/#asterisk InHisName (~InHisName@96.25.37.57) |
| 13:03.32 | Katty | drags in |
| 13:03.49 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:07.42 | InHisName | I am trying to capture the number in "Found peer 'number'" to an asterisk variable, but I haven't discovered how, yet. http://paste2.org/p/2299567 has a snip from debug showing where that line comes from and some outputs of dial functions I was trying out to see if I could bump into that 'number' ? No luck so far. |
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| 13:10.22 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:10.49 | tompaw | [TK]D-Fender: that's what I'm moaning about. There should be an option to kill ALL CALL from a ConfBridge |
| 13:10.49 | tompaw | ALL CALLS even |
| 13:12.35 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 13:12.46 | [TK]D-Fender | tompaw, What does CLI tell you? |
| 13:12.55 | [TK]D-Fender | tompaw, What about AMI? |
| 13:13.55 | *** join/#asterisk Champi (Champi@rootshell.fr) |
| 13:15.09 | Katty | is embarassed to be from 'merica this morning |
| 13:15.24 | Katty | decides she will be an honorary candian for the day. |
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| 13:23.26 | *** join/#asterisk ASUchander (~asuchande@cpe-071-070-226-197.nc.res.rr.com) |
| 13:24.08 | ASUchander | Hi, I'm trying to write an exten rule that matches all inbound calls on a trunk. Can I use exten => *, ... (where the * matches any inbound did?) |
| 13:24.50 | anfecora | try exten => _X., |
| 13:25.34 | anfecora | or exten => _XXXXXXXXXX, |
| 13:25.48 | anfecora | that willl take any inbound did |
| 13:25.52 | ASUchander | thanks |
| 13:26.06 | anfecora | dont forget the _ |
| 13:26.11 | anfecora | before the wildcard |
| 13:26.13 | ASUchander | gotcha :-) |
| 13:26.37 | leifmadsen | ASUchander: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics.html |
| 13:26.38 | *** join/#asterisk biomorph (~biomorph@91.85.204.16) |
| 13:26.51 | leifmadsen | specifically the pattern matching section |
| 13:27.10 | anfecora | Leif your book is 3rd edition now ? |
| 13:27.13 | anfecora | wow |
| 13:27.19 | leifmadsen | has been for nearly a year :) |
| 13:27.29 | anfecora | it has been really long a go |
| 13:27.30 | leifmadsen | working on 4e now |
| 13:27.33 | anfecora | since i meet you |
| 13:28.05 | anfecora | you were just starting hte documentation now is a 3rd edition |
| 13:28.11 | leifmadsen | ah gotcha |
| 13:28.12 | anfecora | congrats didn't know |
| 13:28.16 | leifmadsen | yep been out since April I think |
| 13:28.31 | leifmadsen | errr... no, longer than that :) |
| 13:28.35 | leifmadsen | last April or something |
| 13:28.42 | leifmadsen | I can't remember, it all blurs together |
| 13:28.53 | anfecora | i remember when i took my dcap we did it on paper |
| 13:29.05 | anfecora | and second time you guys were experimenting a software |
| 13:29.09 | anfecora | with steve sokol |
| 13:29.21 | leifmadsen | yeppers |
| 13:29.24 | leifmadsen | that was many moons ago |
| 13:29.50 | anfecora | yup |
| 13:31.12 | anfecora | yup in seatle |
| 13:31.17 | anfecora | i remember jajaja |
| 13:31.26 | anfecora | we used to make fun of mr vyki |
| 13:31.30 | anfecora | i forgot his name |
| 13:31.42 | anfecora | he teamed up twith steve to start the bootcamps |
| 13:31.49 | anfecora | but it was a great experience lol |
| 13:31.59 | anfecora | too long ago |
| 13:33.44 | anfecora | aaa i remember Olle |
| 13:37.17 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-195-88.red.bezeqint.net) |
| 13:39.30 | carrar | Seattle Rocks! |
| 13:39.31 | ASUchander | in asterisk "exten" lines, what does "n" mean (for the 2nd column - after the number)? |
| 13:39.38 | carrar | Other thenthe mayor |
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| 13:40.14 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 13:40.30 | carrar | ~book |
| 13:40.31 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 13:40.33 | ASUchander | is there a way to overwrite a sip header on an inbound call? |
| 13:41.37 | tompaw | [TK]D-Fender: CLI and AMI are the way I do it now, but I just think it's stupid. |
| 13:41.59 | tompaw | [TK]D-Fender: there should be a built-in feature, which (for example) kills all calls within a ConfBridge when ONE designated participant leaves it. |
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| 13:43.55 | anfecora | ASUchander read this http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 13:44.51 | anfecora | AMI is great not stupid |
| 13:45.21 | anfecora | specially wit a goo agi server it rocks |
| 13:46.12 | [TK]D-Fender | tompaw, https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 |
| 13:46.32 | [TK]D-Fender | tompaw, wait_marked yes/no Sets if the user must wait for another marked user to enter before joining the conference. By default, no. |
| 13:46.32 | [TK]D-Fender | end_marked yes/no If enabled, every user with this option in their profile will be removed from the conference when the last marked user exists the conference. |
| 13:46.34 | [TK]D-Fender | ^^^ |
| 13:46.41 | [TK]D-Fender | tomSeems to be an option right there |
| 13:47.18 | [TK]D-Fender | AMI has nothing to do with AGI..... |
| 13:47.56 | anfecora | AMI allows you to manage asterisk |
| 13:48.14 | Dovid | is there any way of doing Set(A=1,B=2) ? |
| 13:48.32 | anfecora | sip headers http://www.voip-info.org/wiki/view/Asterisk+cmd+SipAddHeader |
| 13:49.59 | kaldemar | Dovid: yes. "core show application MSet" |
| 13:50.18 | ASUchander | anfecora: I'm trying to do this on an inbound call.. |
| 13:50.55 | Katty | gooses Qwell |
| 13:52.13 | [TK]D-Fender | <ASUchander> is there a way to overwrite a sip header on an inbound call? <- no |
| 13:52.41 | anfecora | this is a great asterisk version Connected to Asterisk UNKNOWN__and_probably_unsupported currently running |
| 13:52.56 | *** join/#asterisk slav3_kitten (~kitten@unaffiliated/slav3-kitten/x-0866809) |
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| 13:58.24 | tompaw | [TK]D-Fender: this is EXACTLY what I needed! |
| 13:58.35 | tompaw | I don't understand how I missed that, I read the whole page with options :/ |
| 13:58.58 | [TK]D-Fender | tompaw, Not very well... there were multiple lines referring to this functionality |
| 14:00.06 | tompaw | [TK]D-Fender: anyway, thanks a lot. |
| 14:03.39 | *** join/#asterisk fergus (~fergus@178.124.149.113) |
| 14:06.20 | tompaw | OK, so it seems like I need 2 join-contexts, one that marks users, and the other one that doesn't. |
| 14:07.11 | mcolombo | anybody say if there is a variable that contain the number that do Call Forwarding? |
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| 14:10.14 | [TK]D-Fender | mcolombo, What is ... "call forwarding"? |
| 14:10.22 | [TK]D-Fender | mcolombo, This isn't an * feature.... |
| 14:12.58 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:14.02 | mcolombo | Call forwarding (or call diversion), in telephony, is a feature on some telephone networks which let an incoming call to a called party to be redirected to a third party. |
| 14:14.22 | mcolombo | i must have the called party number |
| 14:15.02 | WIMPy | Look at the functions CALLERID and REDIRECTING. |
| 14:15.15 | [TK]D-Fender | mcolombo, Where di I see this as an Asterisk feature? |
| 14:15.57 | tompaw | [TK]D-Fender: it works! \o/ |
| 14:16.02 | mcolombo | i enable it, with follow this link : http://www.voip-info.org/wiki/view/Asterisk+call+forwarding |
| 14:16.11 | [TK]D-Fender | tompaw, OMG, it's almost like it acts as documented! |
| 14:16.19 | tompaw | [TK]D-Fender: exactly, right? |
| 14:16.28 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
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| 14:16.39 | mcolombo | WIMPy : Thanks, i'll try |
| 14:16.40 | [TK]D-Fender | mcolombo, That is DIALPLAN. That isn't an ASTERISK feature... this is YOUR dialplan. |
| 14:16.55 | WIMPy | mcolombo: If you implement it in your dialplan, you have to set the valuues yourself. |
| 14:17.16 | [TK]D-Fender | mcolombo, And now devalidates WIMPy's answer to you |
| 14:18.01 | [TK]D-Fender | And even that dialplan sample is ancient Asterisk 1.0 junk |
| 14:18.35 | WIMPy | Yes, the CF bit of my dialplan is pretty long :-( |
| 14:20.55 | mcolombo | [TK]D-Fender : ok, sorry for my error |
| 14:21.04 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 14:21.27 | mcolombo | I do not know where to look if i do not follow this guidance |
| 14:22.06 | [TK]D-Fender | mcolombo, And you showed a link to a WIKI page with a ton of bad examples and we don't see what YOU have actually done |
| 14:22.44 | [TK]D-Fender | mcolombo, Show that and then we can advise you |
| 14:23.01 | mcolombo | [TK]D-Fender: ok, one second |
| 14:23.30 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 14:28.28 | mcolombo | this is my dialplan : http://pastebin.com/cv1FZqti |
| 14:29.04 | mcolombo | i have copied 2 dial command, is my error |
| 14:32.27 | InHisName | I am trying to capture the number in "Found peer 'number'" to an asterisk variable, but I haven't discovered how, yet. http://paste2.org/p/2299567 has a snip from debug showing where that line comes from and some outputs of dial functions I was trying out to see if I could bump into that 'number' ? No luck so far. |
| 14:33.41 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
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| 14:35.54 | WIMPy | mcolombo: I don't really know where to start. You et an isea from that example on how to use AstBD to store configuration. The rest is rather questionable, and I'd recommend you make up your mind |
| 14:36.09 | WIMPy | on what exactely you want instead of looking for samples. |
| 14:37.37 | [TK]D-Fender | exten => _X.,n,Set(CALLERID(num)=${CALLERID(num)}) <- does nothing |
| 14:38.10 | [TK]D-Fender | exten => _X.,n,Macro(dial-result,${DIALSTATUS}) <- you haven't even DIALED before this |
| 14:38.11 | WIMPy | And Asterisk takes care of the headers itself. |
| 14:38.43 | [TK]D-Fender | exten => _X.,1,MSet(AssertedIdentity="<sip:+39${CALLERID(num)}@pippo.it\;user=phone>") <- you almost certainly shouldn't have quotes around these |
| 14:39.34 | [TK]D-Fender | exten => _*21*X.#.,1,Set(DB(CFIM/${CALLERID(num)})=${REPLACE(EXTEN:4,*#-)}) <- putting anything after the first "." in a pattern is USELESS. and ignored. uo also can't have a pattern saying "ends with". |
| 14:39.41 | [TK]D-Fender | This is all VERY broken. |
| 14:40.48 | mcolombo | wow, great |
| 14:42.00 | mcolombo | Unfortunately, we are doing this thing hurry |
| 14:42.13 | mcolombo | I'll fix it for good |
| 14:42.14 | WIMPy | No way. |
| 14:42.23 | WIMPy | You don;t hurry such things with Asterisk. |
| 14:42.38 | WIMPy | Not if you want something working anyway. |
| 14:43.48 | [TK]D-Fender | exten => #21#,1,DBdel(CFIM/${CALLERIDNUM}) <-- using Asterisk 1.0 vars again.... |
| 14:44.17 | mcolombo | my problem is that i must finish this work as soon as possible.. |
| 14:44.19 | mcolombo | we are doing a test of interconnection and we have a very close deadline |
| 14:44.20 | [TK]D-Fender | exten => _X.,n(cfbs),Set(temp=${DB(CFBS/${EXTEN})}) |
| 14:44.20 | [TK]D-Fender | exten => _X.,n(cfim),Set(temp=${DB(CFIM/${EXTEN})}) |
| 14:44.29 | [TK]D-Fender | Those occur back-to-back and run over each other |
| 14:44.32 | [TK]D-Fender | Completely broken |
| 14:44.38 | WIMPy | Why do you comment on it? You already said it VERY broken. That says it all. |
| 14:45.06 | [TK]D-Fender | mcolombo, Guess you better develop a proper programmer's mindset real fast then ... or hire a consultant. |
| 14:45.06 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:45.11 | WIMPy | It's broken beyond repair. |
| 14:45.20 | [TK]D-Fender | No, it can be repaired... |
| 14:45.38 | [TK]D-Fender | Where "repaired" means "practically rewritten in its entirety" :) |
| 14:45.50 | WIMPy | Sure. But starting from scratch might be easier. |
| 14:45.59 | WIMPy | yeah |
| 14:46.51 | mcolombo | I agree with you, will be done like crap, but for the moment work |
| 14:47.19 | mcolombo | I have to finish these tests then I will have plenty of time to do it again for good |
| 14:47.45 | mcolombo | the problem is as I said before, I have a very close deadline |
| 14:47.52 | [TK]D-Fender | mcolombo, Hire a consultant |
| 14:48.24 | mcolombo | I would, but my boss considers them a waste of money |
| 14:49.56 | [TK]D-Fender | mcolombo, Well you don't seem to know what you're doing, and you have a deadline. SOMEBODY is highly unrealistic here. |
| 14:50.17 | *** join/#asterisk bananapie (~david@186.94.static.oricom.ca) |
| 14:51.03 | mcolombo | secure, but are those things they tell you to do and what should be done |
| 14:53.18 | [TK]D-Fender | mcolombo, My condolences for having an unrealistic boss. |
| 14:53.48 | mcolombo | Thanks |
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| 14:57.12 | jmetro | if you need a patch, re-write what you need statically |
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| 15:04.29 | GeoGeek | Any mondorescue experts here? I am trying a selective restore of my /etc/asterisk and /etc/dahdi directories and it appears to work, but when I reboot the machine the files haven't been restored. |
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| 15:21.18 | Hive | I have a primary and secondary SIP proxy set on my Aastra phone. I'm trying to make a failover plan for if my primary server goes down. I'm testing this by blocking traffic to my primary server. |
| 15:21.24 | Hive | I see the sip peer register on the backup server when we block traffic. However when we unblock traffic, I can't figure out how to force the peer to unregister from the secondary, and reregister on the primary. |
| 15:21.34 | Hive | Does anyone have any ideas for how to force this "switch back to primary when available"? |
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| 15:22.06 | Hive | (I tried 'sip unregister <peer name>' but it just came back online to the secondary, didnt switch to primary) |
| 15:22.09 | drmessano | Hive: This is internal? |
| 15:22.35 | Hive | The servers are remote, not internal |
| 15:22.55 | drmessano | Use DNS SRV |
| 15:23.41 | Hive | I'm sorry but I'm not sure what you mean |
| 15:24.23 | drmessano | http://www.voip-info.org/wiki/view/DNS+SRV |
| 15:26.38 | rrittgarn | Morning gentlement. Still having an issue with my Cisco SPA509Gs.... I cannot for the life of me get the MWI to light up, same exact settings on the PBX work for Aastra, Polycom, Yealink, and Grandstream phones... Thinking there's something "special" with the cisco... such as cisco ignores RFCs because they are cisco. Anyone have any experience with something like this? |
| 15:36.08 | rrittgarn | *crickets* |
| 15:37.13 | ChannelZ | can't speak for those phones specifically but the Linksys generation SPAs work fine.. |
| 15:37.45 | rrittgarn | I believe that these are the ones that came out after cisco rebadged them to 'cisco' as they are the cheap phones |
| 15:38.31 | ChannelZ | yeah I don't know how much they did to the firmware or why they would break that. |
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| 15:38.50 | bitglue | rrittgarn: are the phones subscribed? (sip show subscriptions) |
| 15:39.17 | bitglue | rrittgarn: if not, do they try? (wireshark / tcpdump / sip debug on) |
| 15:39.34 | bitglue | if they try, how do they compare to the linksys devices? |
| 15:39.36 | ChannelZ | None of mine are subscribed but Asterisk does send them notify messages when the voicemail status changes and they pay attention. |
| 15:39.40 | bitglue | if they don't try, why do you have such crappy phones? |
| 15:40.18 | rrittgarn | not showing a subscription |
| 15:40.27 | bitglue | rrittgarn: what about your linksys devices that are working? |
| 15:40.48 | rrittgarn | they are on a different server (Televantage) |
| 15:40.52 | rrittgarn | but working |
| 15:41.19 | bitglue | so, you are saying that different phones, on a different server work |
| 15:41.37 | bitglue | that's about as useful to your troubleshooting effort as saying that there are phones somewhere in the world that have working MWI |
| 15:41.37 | rrittgarn | yes meaning unrelated i know |
| 15:41.43 | rrittgarn | yeah i know |
| 15:41.47 | rrittgarn | i dont have the linksys phones on this server |
| 15:41.50 | rrittgarn | just all the other brands i listed |
| 15:41.54 | rrittgarn | and then these ciscos not working |
| 15:42.02 | rrittgarn | all on the box in question |
| 15:42.05 | bitglue | so, compare the ciscos to something that is working |
| 15:42.11 | bitglue | what's the difference? |
| 15:42.21 | rrittgarn | well for starters as you pointed out they arent subscribing |
| 15:42.36 | bitglue | and the other devices are? |
| 15:42.40 | rrittgarn | yes the are |
| 15:43.01 | bitglue | ok, then i'd look at the network traffic with tcpdump or sip debug in asterisk and see why |
| 15:43.06 | rrittgarn | if i subscribemwi=yes, that means don't send a sip notify unless they subcribe right? just want to make sure I understand that correctly while I'm troubleshooting. |
| 15:43.52 | bitglue | i don't really know about that option |
| 15:44.03 | bitglue | but i do see something other options with "mwi" in the name and "cisco" in the comments |
| 15:44.31 | bitglue | in sip.conf, that is |
| 15:44.56 | rrittgarn | buggymwi ? i have that set = yes |
| 15:45.50 | [TK]D-Fender | rrittgarn, Show us your peer |
| 15:47.29 | *** join/#asterisk skrusty (~ksrusty@62.252.24.138) |
| 15:48.22 | skrusty | afternoon all |
| 15:48.23 | skrusty | are there any online databases with an API that track things like spam callers, numbers used in fraud etc? and that are free! |
| 15:48.48 | WIMPy | skrusty: Have you tried to ask google? |
| 15:48.52 | skrusty | yes :) |
| 15:49.07 | skrusty | i thought i would cut to the chase and see if people use any here that they can recommend |
| 15:49.14 | WIMPy | I get lots of results when searching for numbers. |
| 15:49.25 | *** join/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
| 15:49.30 | skrusty | random numbers or something specific? :) |
| 15:49.53 | WIMPy | Whatever called me. |
| 15:49.53 | *** part/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
| 15:49.56 | skrusty | the main one i seem to find is HumbugLabs |
| 15:50.13 | skrusty | but it's a paid service, and i want an "API" which i can use to filter such calls |
| 15:50.19 | skrusty | inbound and outbound |
| 15:50.30 | skrusty | i guess my point is, if there isn't one people use, i might try starting one! |
| 15:50.57 | *** join/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
| 15:51.02 | WIMPy | I think api and free don;t go well together. |
| 15:51.09 | skrusty | heh :) |
| 15:51.26 | WIMPy | But you can alwys parse the output. |
| 15:51.45 | skrusty | hmmm |
| 15:52.24 | InHisName | I was trying to capture the number in "Found peer 'number'" to an asterisk variable, but I hadn't discovered how, then. http://paste2.org/p/2299567 has a snip from debug showing where that line comes from and some outputs of dial functions I was trying out to see if I could bump into that 'number' ? Now I have answered it myself. |
| 15:52.31 | InHisName | I found the answer to the above question: It was in my extensions.conf all along, but I had it in small letters only and the function call needed to be in all CAPS. --> exten => s,n,NoOp("sipchaninfo="${SIPCHANINFO(peername)}) Once I ran this with the capitalized function name, I saw: --> -- Executing NoOp("SIP/21567248353786-081f5020", ""sipchaninfo="2152345678") in new stack all done with that question, now. |
| 15:53.02 | leifmadsen | drmessano: +1 |
| 15:55.24 | drmessano | :) |
| 15:59.41 | drmessano | @leifmadsen Was the post deleted? |
| 16:00.34 | leifmadsen | yes |
| 16:00.40 | rrittgarn | [TK]D-Fender: http://pastebin.com/rjAnAczi |
| 16:00.56 | drmessano | Guess my work there was done |
| 16:05.18 | *** join/#asterisk kannan (~chatzilla@123.238.229.114) |
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| 16:07.13 | *** join/#asterisk mobile_gordita (~Robert@66-87-93-59.pools.spcsdns.net) |
| 16:07.25 | drmessano | mmmm I love mobile gorditas |
| 16:12.25 | *** join/#asterisk lorsungcu (~anonymous@75-144-37-241-Minnesota.hfc.comcastbusiness.net) |
| 16:14.34 | *** join/#asterisk deo (~dnepangue@112.198.79.110) |
| 16:15.10 | *** part/#asterisk deo (~dnepangue@112.198.79.110) |
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| 16:31.07 | *** join/#asterisk barbosa2 (~juliano.b@177-069-248-125.static.ctbctelecom.com.br) |
| 16:32.54 | barbosa2 | Hi for all, where I found any information on where I can voice synthesizer with asterisk |
| 16:35.28 | cmendes0101 | Text to speech or pitch changer? |
| 16:35.55 | cmendes0101 | If its not one of those I don't think I have an answer lol |
| 16:38.02 | Qwell | cmendes0101: func_pitchshift |
| 16:38.12 | Qwell | oh, nm |
| 16:39.23 | cmendes0101 | ^ @ barbosa2 |
| 16:39.49 | *** join/#asterisk vlad_starkov (~vlad_star@77.41.88.210) |
| 16:40.28 | barbosa2 | cmendes0101, Text to speech |
| 16:40.46 | cmendes0101 | I like cepstral |
| 16:40.48 | leifmadsen | Festival, Cepstral |
| 16:40.53 | leifmadsen | I like Cepstral too |
| 16:40.56 | leifmadsen | good quality |
| 16:41.03 | leifmadsen | and you can purchase the Allison voice too |
| 16:41.15 | leifmadsen | reasonably priced as well, especially if you pre-generate the prompts |
| 16:41.17 | cmendes0101 | festival is free but low quality |
| 16:41.23 | leifmadsen | cmendes0101: +1 |
| 16:41.37 | leifmadsen | Cepstral is good bang for the buck I find |
| 16:41.39 | *** join/#asterisk lorsungcu_ (~anonymous@184-197-206-82.pools.spcsdns.net) |
| 16:42.13 | cmendes0101 | If your just testing, someone created one that uses google api. Nice since has multilanguage support |
| 16:42.17 | barbosa2 | leifmadsen, cmendes0101 ok |
| 16:43.54 | paulc | +1 for Cepstral. I think it was like $25 or $30 for the Allison voice. We use it all the time to text-to-speech prompts for development and QA testing, before getting the actual voice talent in to record prompts. Works a treat! :-) |
| 16:45.01 | cmendes0101 | thats allisons voice on cepstral? |
| 16:45.24 | leifmadsen | well it's a trained version of Allison's voice |
| 16:45.43 | dwayne | I also use, like, and recommend cepstral w/ Allison |
| 16:45.44 | leifmadsen | cepstral has a method of taking a persons voice, and training their system to speak like that person |
| 16:46.08 | cmendes0101 | yah, I didn't know that. Thought it was just coincidence with the name lol |
| 16:46.12 | leifmadsen | I suspect Allison gets some royalty when you purchase "her voice" as well, but I can't verify that |
| 16:46.27 | leifmadsen | oh no, it's a specially trained voice to match that of Allison in Asterisk |
| 16:46.44 | leifmadsen | it's so your text-to-speech voice matches the pre-recorded prompts |
| 16:46.52 | leifmadsen | gives a nice matching system |
| 16:46.57 | cmendes0101 | cool |
| 16:46.59 | *** join/#asterisk angryuser_laptop (~angryuser@143.86.112.78.rev.sfr.net) |
| 16:47.58 | Katty | peeks in |
| 16:48.36 | dwayne | covers Katty's eyes |
| 16:48.48 | Katty | :< |
| 16:48.57 | Katty | i are all grown ups sir! |
| 16:49.14 | dwayne | offers a beer |
| 16:49.32 | Qwell | dwayne: Did you bring enough for the whole class? |
| 16:49.34 | Katty | is it a... craft...beer?! |
| 16:49.52 | Katty | or is it douche beer. |
| 16:49.57 | Katty | douche boy beer. |
| 16:50.09 | Katty | cause i just happen to be a beer snob :< |
| 16:50.36 | paulc | cracks open a Stella |
| 16:50.40 | paulc | (fuck, if only!) |
| 16:50.40 | leifmadsen | gross |
| 16:50.42 | leifmadsen | :) |
| 16:50.45 | leifmadsen | paulc: scotch@! |
| 16:51.24 | Katty | shipyard pumpkin ale. |
| 16:51.36 | paulc | leifmadsen: haha haven't quite got that sophisticated yet.. (I don't drink wine either - go figure).. I smell scotch and think "varnish remover?" - reminds me of my dead french-polishing uncle and his work shop full of solvents.. |
| 16:51.48 | Katty | or maybe a caramel apple spice sam adams |
| 16:51.54 | paulc | Of course - I probably just haven't had the RIGHT scotch yet.. some edumacation needed methinks.. |
| 16:52.02 | Katty | with a slice of tasty maine pumpkin bread |
| 16:52.08 | leifmadsen | paulc: ya depends on the scotch a lot -- I didn't drink wine until I drank it for a week at someones place :) |
| 16:52.40 | Katty | http://42ndrecipestreet.blogspot.com/2012/06/downeast-maine-pumpkin-bread.html <- WARNING: not safe if you are hungry. |
| 16:52.45 | dwayne | Katty, its Georgian beer called Kazbeghi |
| 16:52.59 | Katty | what kind of beer. |
| 16:53.00 | dwayne | from the country, not the state |
| 16:53.05 | Katty | is it a cold brew |
| 16:53.07 | Katty | or a warm brew |
| 16:53.12 | dwayne | yes and delicious too |
| 16:53.15 | dwayne | cold |
| 16:53.16 | paulc | leifmadsen: *nods* I never used to drink beer.. until I did.. guess it's just that acclimitisation.. |
| 16:53.31 | Katty | cold brew...hmm ok |
| 16:53.36 | Katty | what's the after taste like? |
| 16:53.41 | dwayne | Qwell, is it early enough for the whole class to drink? |
| 16:53.41 | Katty | and, give it a bitter factor. |
| 16:53.44 | Katty | on a scale of 1 to 10 |
| 16:53.54 | Qwell | dwayne: It's early enough for me to drink. |
| 16:54.00 | paulc | cold brew vs warm brew - are we talking the brewing process? or serving temperature (in which case warm is just.. wrong) |
| 16:54.08 | Katty | brewing process. |
| 16:54.20 | dwayne | Katty, 3, your lips will kind of smack |
| 16:54.25 | leifmadsen | paulc: yep pretty much same thing. I think there is just a lot of strong flavours going on, so until you kind of get acclimated to them, it's just a lot going on and tastes weird :) |
| 16:55.09 | paulc | leifmadsen: wine tastes "sour" to me.. red, white, whatever - tastes "wine-like" and icky. I'll grow up one of these days.. and become a classy wine drinker.. for now, mine's a Stella :) |
| 16:55.20 | Katty | yes i agree on the wines. |
| 16:55.27 | Katty | for now, i stick to dessert wines. |
| 16:55.37 | Katty | or boones farm, which doesn't actually count as wine. |
| 16:55.54 | Katty | paulc: some beers actually taste a bit better warm |
| 16:56.09 | Katty | and by 'warm' i mean more like room temp |
| 16:56.18 | Katty | not hot chocolate warm |
| 16:56.26 | leifmadsen | paulc: haha, well I love beer, I just think Stella tastes kind of skunky :) |
| 16:56.46 | file | hugs paulc |
| 16:56.47 | leifmadsen | Katty: coffee warm?! |
| 16:56.50 | paulc | Katty: Seems wrong in my head. Maybe cos I've always drunk it cold. |
| 16:56.51 | Katty | those frosty cold lagers just numb the taste buds |
| 16:56.52 | file | RANDOMLY |
| 16:57.02 | paulc | eyes file sideways |
| 16:57.12 | Katty | paulc: marketing pushes cold beer, in frosted mugs. |
| 16:57.18 | leifmadsen | paulc: that's good -- never look at him directly |
| 16:57.35 | leifmadsen | <3's cold beer in frosty mugs |
| 16:57.59 | Katty | most of the full body beers... |
| 16:58.03 | Katty | such as porters. |
| 16:58.11 | Katty | especially the roasty tasting ones... |
| 16:58.14 | Katty | are much tastier a bit warmer. |
| 16:58.21 | *** join/#asterisk nW44b (~Schnitzel@unaffiliated/benwa) |
| 16:58.29 | Katty | i've also noticed pumpkin ales taste a good bit more like pumpkin as they warm to room temp |
| 16:58.39 | Qwell | Katty: yes |
| 16:58.45 | Qwell | Katty: speaking of - what's a good one? |
| 16:58.50 | Katty | shipyard pumpkin ale |
| 16:58.52 | Qwell | or, decent and available, rather |
| 16:58.52 | paulc | I like frosted mugs - they're classy (even at A&W? perhaps?) |
| 16:59.05 | Katty | Qwell: http://www.murphystaproom.net/files/beer/shipyard_pumpkinhead.jpg |
| 16:59.08 | dwayne | Qwell, are we allowed to have pumpkin beer in Alabama ? |
| 16:59.08 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 16:59.09 | Katty | Qwell: good luck finding it tho. |
| 16:59.17 | Qwell | dwayne: probably not |
| 16:59.20 | Qwell | (yes) |
| 16:59.24 | Katty | Qwell: schnucks had it for ONE week. ONE |
| 16:59.26 | Katty | and then GONE |
| 16:59.43 | Katty | paulc: frosted mugs are fun, just keep in mind they will numb your tongue |
| 16:59.51 | Katty | paulc: and you won't taste...much |
| 16:59.54 | *** join/#asterisk felipealmeida (~user@querubim.tecgraf.puc-rio.br) |
| 16:59.55 | file | keep it on topic, all of you! |
| 17:00.07 | Katty | gives file a beer |
| 17:00.15 | file | Katty, I do not drink beer. |
| 17:00.20 | Katty | oh? |
| 17:00.23 | dwayne | file only funnels beer |
| 17:00.24 | Katty | what do you drink then? |
| 17:00.38 | Katty | besides orange juice, obviously. |
| 17:00.40 | file | I drink sugary alcoholic beverages |
| 17:00.46 | Katty | bunny foofoo drinks |
| 17:00.50 | Qwell | file: Don't make me change the topic. |
| 17:00.54 | Katty | i am a fan too. |
| 17:01.06 | Katty | file: have you had a bob marley? it is pineapple juice and coconut rum. |
| 17:01.15 | Katty | file: very sugary. very tasty. |
| 17:01.31 | file | Katty, I think so! I went to LinuxCon in Vancouver and the party had an open bar, so I just had them make me random stuff |
| 17:02.18 | leifmadsen | hopes astricon has scotch at the open bar |
| 17:02.20 | leifmadsen | :D |
| 17:02.21 | file | pretty sure they gave me one |
| 17:04.28 | paulc | hands file a Mike's Hard Lemonade |
| 17:04.35 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 17:04.51 | paulc | word at LinuxCon was that file was harsh flirting with all the bar staff ;-) |
| 17:05.36 | leifmadsen | hands file a hard sarsaparilla |
| 17:05.47 | file | I just want... lunch! |
| 17:06.12 | Katty | i can't picture file flirting with the baristas. |
| 17:06.20 | Katty | that thought makes my brain hurt. |
| 17:06.30 | Katty | unless file's personality has drastically changed since we last hung out. |
| 17:06.30 | file | good |
| 17:08.06 | Katty | i would flirt with a barista tho. |
| 17:11.03 | leifmadsen | would watch that |
| 17:11.36 | Katty | you could join me. |
| 17:11.43 | leifmadsen | true dat |
| 17:11.58 | leifmadsen | if only my flirting skills were of a non-sub-par value |
| 17:11.59 | Katty | maybe we could convince Qwell to flirt with the barista. |
| 17:12.02 | Katty | wouldn't that be a sight. |
| 17:12.11 | Katty | practice on your wife. |
| 17:12.14 | Katty | she will appreciate it. |
| 17:12.14 | leifmadsen | Katty: I agree... I encourage that behaviour |
| 17:12.24 | leifmadsen | Katty: y u no astricon?! |
| 17:12.30 | leifmadsen | I agree with file on lunch soon |
| 17:12.31 | Qwell | O.o |
| 17:12.37 | Katty | would you like the long answer, or the short answer? |
| 17:13.09 | Katty | or is that a rhetorical question |
| 17:14.23 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 17:16.56 | *** join/#asterisk Galen (~Galen@rrcs-76-79-170-42.west.biz.rr.com) |
| 17:18.27 | rrittgarn | finding conflicting info on voipinfo... to set a MOH class to random, is it a ',r' after the file path, or is it random=yes? |
| 17:19.10 | Qwell | rrittgarn: Whatever the sample configs say. |
| 17:19.20 | paulc | random=yes sounds more "right" |
| 17:19.37 | Qwell | hint: neither :) |
| 17:21.25 | rrittgarn | sort=random? |
| 17:21.36 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
| 17:22.52 | rcaskey | so if i want anti-social never-talk-to-a-human-ever SIP trunking where do I head ;P |
| 17:23.27 | Katty | tin can and a string. |
| 17:23.39 | Qwell | Katty: sippy cup and string |
| 17:24.45 | *** join/#asterisk mario_261 (~mario@sje26-1-88-127-240-128.fbx.proxad.net) |
| 17:25.05 | mario_261 | Ouf |
| 17:25.09 | mario_261 | hello |
| 17:26.23 | mario_261 | I'm looking for help |
| 17:26.45 | Qwell | ~ask |
| 17:26.46 | 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. |
| 17:28.47 | mario_261 | I managed to run Asterisk, if I start with: /etc/init.d/asterisk start |
| 17:28.47 | mario_261 | Can not connected with my clients, I do not understand, can you help me ? |
| 17:28.47 | mario_261 | However, if I run with Asterisk: asterisk-cvvv, all are OK it works ... |
| 17:29.01 | Maliuta | mario_261: what kind of help? I'm not sure we have any mental health professionals in channel ... although most of us could probably use one ;P |
| 17:29.35 | Maliuta | mario_261: what type of clients? SIP? IAX2? DAHDI? |
| 17:29.41 | mario_261 | SIP |
| 17:29.54 | Rahoul | have you checked your firewall on the asterisk machine? |
| 17:29.59 | Maliuta | mario_261: so did you put entries in sip.conf for them? |
| 17:30.43 | Maliuta | oh yeah, and is NAT involved at all |
| 17:31.00 | mario_261 | Yes for firewall is ok |
| 17:31.27 | mario_261 | I'll put my sip.conf.... |
| 17:31.40 | Maliuta | pastebin.ca |
| 17:31.41 | mario_261 | 1 min |
| 17:31.44 | mario_261 | ok |
| 17:32.01 | mario_261 | I have my own pastebin :p |
| 17:32.24 | Maliuta | <PROTECTED> |
| 17:32.39 | Rahoul | when you asterisk -vvvc and try to connect a sip phone what message do you see in the asterisk console? |
| 17:32.51 | Rahoul | if any at all |
| 17:33.00 | *** join/#asterisk cmendes0101| (~cmendes01@wtnl.corp.tierra.net) |
| 17:33.00 | *** join/#asterisk becca_r (~becca_r@12.25.151.59) |
| 17:37.24 | mario_261 | http://paste.devosi.org/LECZOSER |
| 17:38.53 | mario_261 | it works well with asterisk-vvvc |
| 17:39.10 | Maliuta | mario_261: and what does "sip show peers" say on the CLI? and answer to Rahouls question would be good to |
| 17:39.18 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:39.19 | mario_261 | but not if I run it with: /etc/init.d/asterisk start |
| 17:40.34 | Maliuta | mario_261: what distro? if it's debian(or debian based) check the contents of /etc/default/asterisk |
| 17:41.07 | mario_261 | *CLI> sip show peers |
| 17:41.07 | mario_261 | Name/username Host Dyn Nat ACL Port Status |
| 17:41.07 | mario_261 | test/test 192.168.1.15 D N 5060 Unmonitored |
| 17:41.07 | mario_261 | test-vostro/test-vostro 192.168.1.88 D N 5060 Unmonitored |
| 17:41.07 | mario_261 | freephonie/0954183460 212.27.52.5 N 5060 OK (44 ms) |
| 17:41.07 | mario_261 | 3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline] |
| 17:41.10 | mario_261 | oups |
| 17:41.12 | mario_261 | sorry |
| 17:41.19 | mario_261 | Debian yes |
| 17:41.28 | Maliuta | so the peers are connected |
| 17:41.36 | Maliuta | and the problem is? |
| 17:42.14 | anfecora | don start asterisk with make config |
| 17:42.25 | anfecora | it hasn't been working |
| 17:42.31 | anfecora | check how many process you have |
| 17:42.40 | anfecora | that happens to me lately |
| 17:42.50 | anfecora | if you check you will see more than 2 asterisk running |
| 17:43.01 | Maliuta | mario_261: stable, unstable, or testing? |
| 17:43.10 | mario_261 | stable |
| 17:43.30 | Maliuta | mario_261: and have you got the context "home" in your extensions? |
| 17:44.13 | mario_261 | I have that one process run... |
| 17:44.51 | mario_261 | Maliuta: extensions.conf |
| 17:46.11 | Maliuta | mario_261: or extension.ael (I prefer to use ael for mine) |
| 17:46.47 | mario_261 | http://paste.devosi.org/s5Y9rEpL |
| 17:46.58 | Maliuta | mario_261: you will need to be careful with the extensions files, the default debian setup has all kinds of oddness in it |
| 17:47.53 | Rahoul | mario_261: When you say the sip client won't connect, you mean it won't register or it won't allow you to make calls? |
| 17:48.23 | mario_261 | client won't connect |
| 17:48.48 | Rahoul | well per your show sip peers freephonie seems to be registered |
| 17:49.14 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 17:49.25 | mario_261 | when I start with /etc/init.d/asterisk start it does not connect to server |
| 17:49.38 | Rahoul | well it wont |
| 17:49.46 | Rahoul | that will only get asterisk running in the background |
| 17:49.55 | Rahoul | then do /usr/sbin/asterisk -vvvr |
| 17:50.01 | Rahoul | and you will connect to asterisk CLI |
| 17:50.09 | Maliuta | mario_261: from the cli do a "dialplan show" |
| 17:52.14 | Maliuta | mario_261: it might also be worth doing a comparison of how the processes are running (i.e. compare "ps aux | grep asterisk" from when it's started by hand and by the init script) |
| 17:54.12 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CMb89bc9d2e1a5.cpe.net.cable.rogers.com) |
| 17:54.20 | *** join/#asterisk vlad_starkov (~vlad_star@77.41.88.210) |
| 17:54.50 | mario_261 | I'll eat, back in 15 min thank you |
| 17:56.49 | *** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca) |
| 17:58.16 | *** part/#asterisk GeoGeek (~steve-o_@12.71.122.227) |
| 17:58.42 | nny | how does IAX2 waulify? ping or app layer? I have an issue where I can't waulify a host behind NAT and I have the port forwarded |
| 17:59.45 | nny | w/q lol |
| 17:59.54 | nny | I sound like barabra walters |
| 18:00.59 | *** join/#asterisk Neptu (~Neptu@c213-89-2-159.bredband.comhem.se) |
| 18:06.29 | *** join/#asterisk mcrane (~mcrane@24-117-99-123.cpe.cableone.net) |
| 18:15.46 | *** part/#asterisk mcrane (~mcrane@24-117-99-123.cpe.cableone.net) |
| 18:17.38 | jaytee | is Asterisk 1.8 no longer available in the repos on epel? |
| 18:21.10 | *** join/#asterisk ThomCalvin (~ThomCalvi@69.198.193.49) |
| 18:31.09 | ThomCalvin | Newbie Here.. my small business was given a Switchvox Digium/Asterisk AA300. The previous owner of the box has software loaded with all previous contacts.. and there's several errors upon startup. Would I better served to try and "decode" what they've done or load fresh software onto the box? |
| 18:31.52 | pabelanger | ThomCalvin: your best to contact Digium directly, most of us don't know anything about Switchvox |
| 18:32.08 | ThomCalvin | Good to know, thx. |
| 18:32.23 | pabelanger | I'm sure you can factory reset the device, but not sure how to do it |
| 18:40.24 | *** join/#asterisk j-fish (~hhkkhkj@unaffiliated/j-fish) |
| 18:41.47 | j-fish | Any ideas how to fight a spammer?confernceing calls between my office and another business and he's spoofing the caller id to match those other businesses |
| 18:43.13 | paulc | Generally we send the boys round to "have a word" |
| 18:43.33 | j-fish | i have no idea who's doing this. |
| 18:49.39 | *** join/#asterisk w9sh (~chatzilla@64.238.96.125) |
| 18:53.45 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 18:56.02 | rrittgarn | So just FYI to anyone who was looking at my cisco problem... I removed the subscribemwi completely... and that seems to do it... what a pain... |
| 18:56.25 | mario_261 | re |
| 18:57.27 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 18:57.28 | mario_261 | au cas ou je me donne une chance de mieux me faire comprendre pour mon problème |
| 18:57.42 | *** join/#asterisk hariom (~hariom@117.195.184.72) |
| 18:57.43 | mario_261 | parmi vous quelqu'un parle français ? |
| 18:58.18 | [TK]D-Fender | Peut-etre qu'une dixaine.... |
| 18:58.21 | mario_261 | someone speak French? |
| 18:58.27 | mario_261 | ha c'est déja pas mal ;) |
| 18:59.12 | mario_261 | car je sais écrire quelque mot anglais mais c'est tip top... |
| 19:00.20 | mario_261 | I can explain my problem in French [TK]D-Fender ? |
| 19:00.21 | hariom | Which codec provides the lowest bit rate possible and can run on Asterisk? |
| 19:00.37 | *** join/#asterisk vlad_starkov (~vlad_star@77.41.88.210) |
| 19:00.39 | [TK]D-Fender | marOK, j'ai pas entierment retracee vos questions. T'es rendu ou finallement? |
| 19:01.10 | jpsharp | hariom: ilbc or LPC10. |
| 19:01.26 | mario_261 | en faite je comprend pas trop ce qu'il voulais essayé de me dire.... |
| 19:01.33 | [TK]D-Fender | mario_261, Ton dialplan et tellement <<basic>> et meme qui'il y-a des echecs, ca pourrait-etre fonctionelle cependant |
| 19:01.55 | [TK]D-Fender | mario_261, <<qu'il>> ? Qui? |
| 19:02.15 | mario_261 | en gros je me suis configuré un petit serveur Asterisk avec ce tuto http://artisan.karma-lab.net/comment-mettre-place-sa-telephonie-grace-a-asterisk |
| 19:02.31 | [TK]D-Fender | hariom, Don't do it ... LPC = Domo Arigato .... |
| 19:02.31 | hariom | jpsharp: Is ilbc less than g729? Is there any chart like think to see the possible bit rates of various codec? |
| 19:02.45 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 19:03.05 | mario_261 | et il disent d'exécuter Asterisk en mode débug avec cette commande : asterisk -cvvv |
| 19:03.39 | mario_261 | donc moi depuis le début je fait comme ça et ça fonctionne j'ai le résulta que je souhaite "enfin a peut prêts" |
| 19:03.48 | [TK]D-Fender | mario_261, No.. ce maniere de le lancer c'est pour le gere en mode interactive et <<live>> |
| 19:03.53 | *** join/#asterisk bullium (~will@216.68.250.27) |
| 19:04.00 | [TK]D-Fender | mario_261, C'est a dire que en quitter .... il QUITTE |
| 19:04.10 | *** join/#asterisk m0spf (~steve@2001:ba8:1f1:f12e::2) |
| 19:04.31 | mario_261 | donc viens le moment ou je souhaite lancer le serveur normalement avec cette commande : /etc/init.d/asterisk start |
| 19:04.40 | [TK]D-Fender | mario_261, C'est un affaire a faire quand tu devrais determiner pourquoi il reussi pas de lancer en mode <<daemon>> |
| 19:04.46 | mario_261 | et la paff mes client SIP ne ce connecte plus |
| 19:04.50 | bullium | can asterisk's voicemail system work without using the voicemail.conf file say by sip.conf and extentions.conf only? |
| 19:04.57 | mario_261 | j'utilise "Jitsi" comme client |
| 19:05.46 | [TK]D-Fender | oublie le client. Est-tu dans console d'asterisk? |
| 19:05.53 | mario_261 | en gros voila mon problème "en tout cas l'un de mes problème" :p |
| 19:05.57 | mario_261 | oui j'y suis |
| 19:06.22 | [TK]D-Fender | faire "exit" |
| 19:06.44 | mario_261 | *CLI> exit |
| 19:06.44 | mario_261 | No such command 'exit' (type 'core show help exit' for other possible commands) |
| 19:06.44 | mario_261 | *CLI> |
| 19:07.07 | mario_261 | pour coupé d'habitude je CTRL C |
| 19:07.18 | mario_261 | exit ne veut pas |
| 19:07.21 | jpsharp | hariom: You're right, g729 is lower than ilbc. |
| 19:07.55 | [TK]D-Fender | marAlor tu l'a lancer avec gc... ne le fait plus de meme |
| 19:08.12 | [TK]D-Fender | mario_261, faire avec le init script. |
| 19:08.48 | mario_261 | /etc/init.d/asterisk start comme ça ? |
| 19:09.03 | mario_261 | /etc/init.d/asterisk -cvvv start ? |
| 19:09.12 | [TK]D-Fender | premiere |
| 19:09.18 | *** join/#asterisk prg3 (~prg3@chatter.majestik.org) |
| 19:09.22 | jpsharp | hariom: http://www.speex.org/comparison/ |
| 19:09.27 | mario_261 | c'est fait |
| 19:09.29 | jpsharp | Compares a bunch of codecs. |
| 19:09.30 | mario_261 | il es lancer |
| 19:09.41 | mario_261 | par contre mes client SIP ne se connecte plus comme ça.... |
| 19:09.54 | jpsharp | I think speex is about as low as you can get and have audio that doesn't sound like crap. |
| 19:10.44 | [TK]D-Fender | mario_261, Tu devrais avoir etait retourner au console. En suite faire "asterisk -r" pour connecter |
| 19:11.18 | mario_261 | c'est bon |
| 19:11.34 | mario_261 | je suis bien a CLI> |
| 19:11.42 | [TK]D-Fender | mario_261, Ok, premier etap complete |
| 19:11.50 | mario_261 | je savais pas ça tu vois déja |
| 19:12.13 | mario_261 | les client ce connecte toujours pas par contre... |
| 19:12.47 | [TK]D-Fender | ou est-ce que ils sont? |
| 19:12.59 | mario_261 | sur mes ordi dans le LAN |
| 19:13.31 | mario_261 | les serveur est dans le LAN aussi |
| 19:15.01 | [TK]D-Fender | Alors il ne peut pas appler meme entre eux? |
| 19:15.09 | mario_261 | non |
| 19:15.11 | mario_261 | rien |
| 19:15.31 | mario_261 | le client me renvois qu'il peut pas ce connecté |
| 19:16.05 | mario_261 | les deux client du coup car j'ai deux PC la avec moi |
| 19:16.20 | [TK]D-Fender | Assurer-vous que ton FIREWALL n'est pas en train de tout-bloquee |
| 19:16.41 | mario_261 | il es ouvert |
| 19:16.51 | mario_261 | au 6050 UPD ? |
| 19:16.52 | [TK]D-Fender | iptabless --list |
| 19:16.58 | [TK]D-Fender | BOUCOUP plus. |
| 19:17.07 | [TK]D-Fender | enlele-les entirement. |
| 19:17.14 | [TK]D-Fender | enleve* |
| 19:17.14 | mario_261 | 5060/udp ALLOW Anywhere |
| 19:17.18 | [TK]D-Fender | pour vos teste |
| 19:17.20 | [TK]D-Fender | .... |
| 19:17.27 | [TK]D-Fender | ca n'est pas la VOIX |
| 19:17.44 | mario_261 | je met carrément une plage d'ip ? |
| 19:17.52 | [TK]D-Fender | SIP c'est un protocol d'initiation... le voix ce fais sur un range eleve |
| 19:18.06 | [TK]D-Fender | 10000-20000 UDP normalement |
| 19:18.08 | mario_261 | de 5000 à 5100 ? |
| 19:18.14 | mario_261 | ha oui quand meme |
| 19:18.17 | [TK]D-Fender | rtp.conf <- |
| 19:18.23 | paulc | returns to his screen to find it all in French *blinks* |
| 19:18.30 | Katty | je voudrais une omelette |
| 19:18.39 | mario_261 | pourquoi ça marche avec l'autre commande de démarrage ? |
| 19:18.41 | Katty | et le fromage |
| 19:18.48 | paulc | I just had scrambled eggs at work - management threw a thanksgiving brunch :) WITH BACON |
| 19:18.49 | mario_261 | merci katty :D |
| 19:19.06 | Katty | mmm bacon |
| 19:19.10 | paulc | je voudrais une biere.. mais realment, plus d'une c'est meilleur |
| 19:19.18 | paulc | finishes butchering the french language |
| 19:19.22 | [TK]D-Fender | Katty, How to speak southern :. Word "Omlette". Sample usage "You done did me wrong but omlette you off easy this time!" |
| 19:19.32 | paulc | LOL :-) that's awesome |
| 19:19.33 | mario_261 | et les français ! |
| 19:19.36 | hariom | jpsharp: thanks |
| 19:19.43 | Katty | facepalms. |
| 19:19.44 | jpsharp | no problem. |
| 19:19.51 | mario_261 | savez vous que vous avez une doc a remplir ? |
| 19:19.52 | mario_261 | http://wiki.asterisk-france.org/index.php/Accueil |
| 19:19.53 | Katty | [TK]D-Fender: yes. i had to read it outloud to get it tho. |
| 19:20.01 | paulc | I saw honey boo boo for the first time the other night. My life hasn't been the same since. I'm seriously worried about the species. |
| 19:20.07 | mario_261 | car elle es un peut vide :D |
| 19:20.19 | Katty | paulc: YES. i also made an effort to look it up since it was on reddit and i had /no idea/ what they were talking about. |
| 19:20.22 | Katty | paulc: it was... |
| 19:20.29 | Katty | paulc: uncomfortable... to watch |
| 19:21.05 | Katty | paulc: honestly? i feel bad for the kid |
| 19:21.12 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 19:21.13 | Katty | paulc: she has /no idea/ what's going on.... |
| 19:21.25 | Katty | paulc: reality will catch up with her soon enough...and she will be crushed :< |
| 19:21.30 | *** join/#asterisk angryuser_laptop (~angryuser@85-170-147-158.rev.numericable.fr) |
| 19:21.35 | [TK]D-Fender | mario_261, Alors demontrer qu'il est vide, et en suite (et inclus) faire "sip set debug on" en * CLI et essai de REPARTIR vos soft-phones.... on devrait voir les comms pour... |
| 19:21.37 | [TK]D-Fender | ~pb |
| 19:21.37 | 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. |
| 19:21.39 | [TK]D-Fender | ^^^ |
| 19:22.07 | [TK]D-Fender | mario_261, http://paste.devosi.org <--- debile, utilise pastebin.com en place S-T-P |
| 19:22.18 | mario_261 | Rooooo |
| 19:22.25 | mario_261 | c'est nous qui l'avons fait ! |
| 19:22.34 | Katty | i love the sound of french. |
| 19:22.35 | mario_261 | il es codé par un copain |
| 19:22.43 | mario_261 | et finiolé a mes soin ! :D |
| 19:22.44 | Katty | someone tell me about chemistry in french. |
| 19:22.45 | [TK]D-Fender | mario_261, Recommence de nouveau :p |
| 19:22.56 | Katty | i want to hear about ALL THE PARTICULATES AND ISOTOPES, in french please. |
| 19:23.03 | mario_261 | et surtout il ce trouve chez moi ! sur un serveur Auto Héberger ;) |
| 19:23.07 | *** join/#asterisk qakhan (~qakhan@208.253.91.58) |
| 19:23.09 | qakhan | hi all |
| 19:23.11 | [TK]D-Fender | Katty, That;s just so ripe for double entendres.... |
| 19:23.27 | qakhan | how to install phpagi on linux for asterisk |
| 19:23.42 | [TK]D-Fender | mario_261, defois la merde ne travers pas loin ;) |
| 19:23.50 | mario_261 | LOL |
| 19:23.51 | paulc | Katty: Yeah, it's true. But maybe the few thousand she's got in the bank will make up for it. I dunno. I shake my head. And this was on TLC - doesn't that stand for The LEARNING channel?! |
| 19:23.53 | [TK]D-Fender | qakhan, There is no "install" |
| 19:23.59 | Katty | it's like hearing Junk-Y ramble on for an hour |
| 19:24.00 | mario_261 | pourquoi tu me dit ça ? :D |
| 19:24.08 | jpsharp | qakhan: from phpagi.sourceforge.net? Just untar it and stick it somewhere in your php include path. |
| 19:24.14 | Katty | i don't have a damn clue what that man is saying, but he better keep talking. |
| 19:24.15 | [TK]D-Fender | qakhan, It's a dumb PHP you inclue. Put it wherever you want and INCLUDE it in your own projects |
| 19:24.19 | [TK]D-Fender | include. |
| 19:24.30 | Katty | [TK]D-Fender: you must serenade me for an hour someday. |
| 19:24.46 | [TK]D-Fender | Katty, Nah, he's full-on Quebecois, his has a slightlyy different flavour :) |
| 19:25.11 | qakhan | oh thanks [TK]D-Fender |
| 19:25.13 | Katty | well we're talking about Yum and Yum... |
| 19:25.27 | [TK]D-Fender | Katty, Hit up my FB page and you'll get links to some of my live performances ;) |
| 19:25.36 | Katty | that's not the same, dear. |
| 19:25.43 | Katty | i demand your full attention. |
| 19:26.18 | [TK]D-Fender | mario_261, Que tu fais l'hebergement direct ne dit pas que le produit est bonne ;) En tous-cas, j'attends ton pastebin... |
| 19:26.38 | [TK]D-Fender | Katty, I haven't met even a handful that are capable of handling that. |
| 19:27.08 | [TK]D-Fender | Katty, I would burn you :O |
| 19:27.56 | mario_261 | [TK]D-Fender: comment tu as ouvert tes port pour Asterisk ? |
| 19:28.32 | *** join/#asterisk timholum (~chatzilla@68-117-120-138.static.eucl.wi.charter.com) |
| 19:28.38 | [TK]D-Fender | marJ'ai jamais fermer dans premiere place... |
| 19:29.04 | timholum | is there a way to tell if a call is a transfer in the dialplan? |
| 19:29.25 | *** part/#asterisk keycruncher (~keycrunch@c-174-59-233-118.hsd1.pa.comcast.net) |
| 19:29.27 | [TK]D-Fender | timholum, depends .... there is a certain channel variable you could look for... |
| 19:29.56 | timholum | which veriable is that? |
| 19:30.24 | timholum | or should I set something on inbound calls, and if that is set, then it is a transfer? |
| 19:30.39 | mario_261 | 9a crain rien pour le serveur d'ouvrir une aussi grosse plage de port ? |
| 19:30.51 | [TK]D-Fender | timholum, You should really look at the list of reserved asterisk channel variables.... |
| 19:30.56 | *** join/#asterisk albania-asterisk (~chatzilla@92.60.25.14) |
| 19:31.02 | albania-asterisk | hello there |
| 19:31.06 | timholum | ok, will do, thanks |
| 19:31.35 | mario_261 | [TK]D-Fender: au niveau de la sécurité c'est pas top si ? |
| 19:31.52 | [TK]D-Fender | mario_261, 1 port pas direction de voix par conversation, un autre pour le DTMF, etc. Alors vers 4 par appele actifs en cours |
| 19:32.08 | albania-asterisk | time after time i have "RTP Read too short (32, expecting 36) " |
| 19:32.27 | [TK]D-Fender | mario_261, Le risque securitaire et PAREIL. Si la SERVICE n'est pas securise, le nobre de ports ne fait aucun difference. |
| 19:32.48 | mario_261 | c'est a dire si le service n'est pas sécurisé ? |
| 19:33.00 | mario_261 | comment ça ce sécurise Asterisk ? |
| 19:33.06 | [TK]D-Fender | drmessano, You favourite question about # ports = security VS if listener suck, # of ports = irrelevant :) |
| 19:33.09 | [TK]D-Fender | your* |
| 19:33.15 | albania-asterisk | this i have also before and it happened when one of the peers has packet loss, but from 1 year it has been only 'RTP Read too short" only, without the (x expecting y ) |
| 19:33.30 | mario_261 | j'ai besoin de comprendre car ça m'intéresse j'en es besoin... |
| 19:33.50 | albania-asterisk | i need to know why this new RTP Read too short started to come also with the these parameters, and i am corried |
| 19:34.07 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 19:34.21 | [TK]D-Fender | mario_261, Un port-scanne vas les trouver si ils son 10 ou 10,000. Ci la PROCESSE n'est pas securise, le # de port ne change rien |
| 19:34.36 | mario_261 | ok je comprend |
| 19:34.47 | mario_261 | et Astérisk est bien sécurisé par defaut ? |
| 19:35.06 | albania-asterisk | time after time i have "RTP Read too short (32, expecting 36) " |
| 19:36.45 | albania-asterisk | time after time i have "RTP Read too short (32, expecting 36) " and not just RTP read too short that was coming usually where some ip quality issues with peers, only this ..RTP read too short ( x expecting y |
| 19:37.26 | *** join/#asterisk autofsckk (~que@unaffiliated/autofsckk) |
| 19:38.41 | *** join/#asterisk autofsckk (~que@unaffiliated/autofsckk) |
| 19:38.51 | albania-asterisk | can you tell me what is the difference between between cli msg ""RTP Read too short (x, expecting y) " and "RTP Read too short" |
| 19:39.13 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 19:39.37 | autofsckk | hello, im getting a lot of UNREACHABLE and Lagged messages from my asterisk box, its a vps, im i being attacked or something like that? how can i know? :S |
| 19:39.55 | Katty | [TK]D-Fender: i didn't mean like that. |
| 19:40.04 | Katty | [TK]D-Fender: i mean your complete, undivided attention, while speaking french. |
| 19:40.07 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 19:40.18 | Katty | [TK]D-Fender: i have no interest in your.... other qualities. |
| 19:40.35 | Katty | [TK]D-Fender: which i'm sure are quite lovely. |
| 19:40.59 | albania-asterisk | can somebody help me on telling me what is the difference between cli msg "RTP Read too short (x, expecting y) " and "RTP Read too short" |
| 19:44.07 | *** part/#asterisk j-fish (~hhkkhkj@unaffiliated/j-fish) |
| 19:44.08 | jpsharp | autofsckk: That's typical of network congestion. |
| 19:45.01 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 19:45.26 | albania-asterisk | jsharp can you help me, do you know the differnce between the msg i asked for, what did the developers intend to show |
| 19:48.06 | jpsharp | albania-asterisk: I don't know the difference, sorry. |
| 19:48.09 | autofsckk | jpsharp: and how do i know if im being attacked or hacked? something i can read to see if im exposed? |
| 19:48.46 | jpsharp | If you're being hacked or hack attempt, you'll see a bunch of failed registrations, not lagged or unreachable. |
| 19:49.56 | autofsckk | ok, not failed registrations, just lagged and then unreachable |
| 19:49.57 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 19:50.07 | jpsharp | network or VPS issues. |
| 19:51.15 | autofsckk | ok, thanks jpsharp |
| 19:53.34 | *** join/#asterisk mobile_gordita (~Robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:53.42 | Katty | noms on mobile_gordita |
| 19:54.43 | Katty | :< |
| 19:54.51 | Katty | talk about fast food.... |
| 19:55.19 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:55.50 | Rahoul | autofsckk: i sent you a pm with info about low latency vps I use myself for asterisk inc ase you have problems like these again. |
| 19:56.50 | bitglue | hm so, i have a thing that happens after queue calls (via Dial() with the F option), and somehow one of my queue members has gotten stuck in that routine after his phone crashed. It seems asterisk hasn't noticed, and still thinks the phone is on a call, so he's not getting more calls. Is there something I can do to have asterisk realize that this call is dead? |
| 19:57.20 | bitglue | ie, "core show channels" shows a channel for that SIP endpoint, even though the endpoint crashed and the phone has since been rebooted. |
| 19:59.52 | *** join/#asterisk jaxon007_ (~jaxon007_@14.97.231.215) |
| 20:00.11 | *** join/#asterisk donnib (~donnib@2.105.191.170) |
| 20:01.02 | jaxon007_ | can anyone tell me production experince of asterisk-10?\ |
| 20:02.03 | drmessano | jaxon007_: It works |
| 20:02.36 | drmessano | jaxon007_: Nothings changed since it was suggested to you yesterday |
| 20:02.57 | cmendes0101 | yes I concur |
| 20:06.56 | jaxon007_ | @drmessano: as per your suggestion , i go with asterisk-10 and confbridge ...but i want to know the performance of asterisk-10. any bugs faced on asterisk-10 on production? |
| 20:08.21 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 20:10.17 | cmendes0101 | 10 is a standard release, its not a beta version or anything |
| 20:11.22 | *** join/#asterisk doolittlework (~doolittle@41-134-22-14.dsl.mweb.co.za) |
| 20:11.32 | cmendes0101 | I'm sure theres some bugs like anything. Check out https://issues.asterisk.org/jira/browse/ASTERISK |
| 20:12.00 | hariom | I have installed my * system as: ./configure; make; sudo make install; sudo make config. Then I chagned the ownership to local user. Also made respective changes in asterisk.conf for user and group. I am facing a problem, when I start the server, it doesn't take up the call unless I login to the system. When I grep, I see asterisk running but not safe_asterisk. How to fix that when server restarts, asterisk should automatically pick calls |
| 20:12.01 | doolittlework | how much bandwidth does a g729 call take |
| 20:13.35 | cmendes0101 | doolittlework: 8kbps |
| 20:13.50 | cmendes0101 | doolittlework: http://voiceonbits.com/2010/08/15/codec-bandwidth-calculation-g711g729/ |
| 20:14.45 | doolittlework | cmendes0101 is the codec one can get of the asterisk.hosting.lv the right g729 codec |
| 20:16.54 | cmendes0101 | I'm not familiar with that one but there marking that as G. 729. |
| 20:17.47 | cmendes0101 | doolittlework: There is licensing you need for it tho |
| 20:18.12 | cmendes0101 | doolittlework: check this out http://www.voip-info.org/wiki/view/Asterisk+G.729+Licensing |
| 20:18.18 | doolittlework | i have been using the g729 of that site with no problems |
| 20:18.35 | cmendes0101 | "should" |
| 20:18.43 | cmendes0101 | drop that in somewhere lol |
| 20:18.53 | doolittlework | was just wondering if it is the right codex when one talks about compresions |
| 20:19.19 | *** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca) |
| 20:19.36 | [TK]D-Fender | G.729 = 9.6kpbs payload |
| 20:20.00 | [TK]D-Fender | RTP = 20kbps overhead @ 20ms packetization rate. |
| 20:20.14 | [TK]D-Fender | So typically 29.6 kbps / direction |
| 20:20.27 | doolittlework | thank you |
| 20:25.14 | [TK]D-Fender | packs up to head home... |
| 20:25.36 | *** join/#asterisk AkkerKid (~AkkerKid@50-200-18-202-static.hfc.comcastbusiness.net) |
| 20:26.49 | *** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca) |
| 20:27.47 | AkkerKid | Hey all! I need a GotoIf statement that checks if a string has a colon in it. I can't get it to execute properly. |
| 20:28.01 | AkkerKid | this is what I've got... exten => s,n,GotoIf($[${LEN("${CallerID(name)}"=~"(\:)")} > 0]? |
| 20:28.02 | mario_261 | have you made a port forwarding for asterisk in UFW ? |
| 20:28.38 | AkkerKid | How far off am I? |
| 20:28.39 | mario_261 | is there a documentation about it ? |
| 20:30.21 | Katty | mario_261: can i help you with something? |
| 20:30.26 | Katty | mario_261: perhaps in english? |
| 20:30.28 | *** join/#asterisk angryuser_laptop (~angryuser@85-170-147-158.rev.numericable.fr) |
| 20:31.12 | mario_261 | I'm bad in English :-/ |
| 20:31.59 | mario_261 | but there is a documentation in English I take ;) |
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| 20:34.57 | mathi | hi |
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| 20:35.20 | mathi | I would like to put the content of a variable in an URL that I will use with CURL |
| 20:35.49 | mathi | e.g. http://example.be/test/${MY_VAR} |
| 20:36.05 | mathi | the problem is that MY_VAR may contain characters that will mess up the url, for example a slash |
| 20:36.15 | mathi | how can I url encode a variable? |
| 20:37.43 | cmendes0101 | URIENCODE? |
| 20:40.36 | mathi | ah thanks)) |
| 20:41.21 | *** part/#asterisk Nemus (~Nemus@c-76-27-99-15.hsd1.ut.comcast.net) |
| 20:41.30 | fenrus | URIN-CODE ? |
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| 20:42.33 | dandate2 | is anyone else having problems with callcentric right now? |
| 20:43.37 | drmessano | People use Callcentric? |
| 20:46.35 | cmendes0101 | I dont like them |
| 20:46.56 | cmendes0101 | Kept having problems when I was using them for 1 personal number |
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| 20:53.11 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:53.19 | gavimobile | hey folks, I need some explination here please. sometimes I use a softphone to call my cellphone and than once the call connects I have a call transfer option to enter another number which will conect my cell phone to this number which was entered. in my cdr it shows 1 outgoing call to my cellphone, but the transfer call number shows my main did number as an incomming call. rather than showing the transfer number I entered as an outgoing |
| 20:55.06 | bullium | running version 1.8.11-cert7 and I'm curious, in order to get voicemail working do have to set something in voicemail.conf or can I do everything I need in users.conf |
| 20:58.06 | [TK]D-Fender | Yes you need voicemail.conf and users.conf =trash |
| 20:59.47 | bullium | [TK]D-Fender: cool thats all I needed to know :) |
| 21:00.15 | bullium | quit |
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| 21:18.26 | Amnesia | question, is it possible to let 2 sip clients communicate with one another, without the interference of a pbx? |
| 21:18.46 | pabelanger | Amnesia: yes, just directly dial it |
| 21:18.55 | Amnesia | sweet, thx |
| 21:19.58 | *** join/#asterisk luckman212 (~luckman21@2001:470:8abb:0:211:32ff:fe10:cdc1) |
| 21:20.44 | Amnesia | I'm curious how that's going to act with NAT being issued^^ |
| 21:21.30 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 21:21.49 | AkkerKid | if a macro has a [context] include line just under it, how does the referenced [context] interact with the original macro? |
| 21:23.12 | jpsharp | Amnesia: Just have to make sure you configure Asterisk properly to deal with the nat configuration. |
| 21:23.33 | Amnesia | wel I'm planning not to use asterisk at all:) |
| 21:23.57 | AkkerKid | for instance, what happens in this situation in extensions.conf? http://pastebin.com/6dEKPmNA |
| 21:24.33 | jpsharp | Amnesia: Oh, well, then, i'd imagine it depends on how smart the clients are. |
| 21:24.36 | pabelanger | Amnesia: that is the down side, you now need to manage how all calls will connected to the phone. So, rather then pulling your phone directly on the internet, use a PBX to manage it |
| 21:26.02 | Amnesia | hm, the idea is a replacement for msn for video conferencing.. |
| 21:26.27 | *** part/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 21:28.09 | jpsharp | Well, you could have the clients dial directly to each other by IP, assuming the client supports that as well as dialing without being registered to a sip proxy. Some clients aren't that smart. |
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| 21:28.27 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 21:28.44 | Amnesia | hm, I'll try ekiga |
| 21:29.32 | [TK]D-Fender | AkkerKid: Your firs context there isn't even a macro |
| 21:29.45 | [TK]D-Fender | AkkerKid: [macro-actualnamehere] |
| 21:29.45 | hariom | I have enabled speex in asterisk. How to get speex prompts? By default I have got wav, ulaw, alaw and gsm |
| 21:29.53 | [TK]D-Fender | AkkerKid: {macro] = just a context |
| 21:30.32 | [TK]D-Fender | hariom: Go look for them in speex then |
| 21:30.47 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-tksyibchcemixhxh) |
| 21:30.48 | [TK]D-Fender | And "default" is pretty meaningless |
| 21:32.23 | hariom | [TK]D-Fender: In order to enable speex between client and server, do I need speex encoded prompts? or wav/ulaw will get converted into speex before playing on the channel? |
| 21:32.46 | jpsharp | Asterisk can transcode the prompts. |
| 21:32.48 | hariom | [TK]D-Fender: my client has speex support. |
| 21:33.07 | [TK]D-Fender | hariom: This is transcoding 101. If you play a prompt you don't have in the format of the call, * will transcode from whatever format you do have |
| 21:33.23 | hariom | jpsharp: I see, so then I don't have to worry about speex prompt. |
| 21:34.00 | jpsharp | No, but if you have a heavy call volume, it would be better to provide native prompts. |
| 21:34.40 | hariom | jpsharp: Then how to get speex encoded prompts? Can sox do that? |
| 21:35.00 | hariom | jpsharp: I have .wav files |
| 21:35.15 | AkkerKid | [TK]D-Fender: I have a FreePBX generated macro that I want to substitute with my own. When I copy-paste the content of that macro under the name of the include statement in the original, stuff doesn't work right... |
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| 21:35.51 | [TK]D-Fender | AkkerKid: and you are still talking in generic terms like ... stuff.... |
| 21:36.05 | [TK]D-Fender | AkkerKid: The details is what you've messed up |
| 21:36.48 | AkkerKid | this is what I'm working with http://pastebin.com/h2EpRiHC |
| 21:37.38 | AkkerKid | basically I've added a couple lines to this macro but it gets rewritten when I update through FreePBX so I want me few extra lines to be available by the include statement instead |
| 21:37.51 | AkkerKid | and use my code instead of the default stuff |
| 21:38.30 | AkkerKid | all my code changes is if a callerid comes in with a colon in it, it'll skip any further manipulation and exit. |
| 21:39.04 | [TK]D-Fender | AkkerKid: First modding the file that appears in is a shoot-on-sight-offence |
| 21:39.10 | hariom | I have recorded wav files and they are 8Khz 32bit floating point. They play well in * but when I exported them via audacity to 8Khz 16bit, it doesn't play well in *. I only hear white noise like sound |
| 21:40.00 | AkkerKid | [TK]D-Fender: understood. so i'm putting my edited version into extensions_custom.conf instead of extensions_additional.conf |
| 21:40.18 | AkkerKid | that works for all of my other custom stuff |
| 21:42.04 | jpsharp | quit |
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| 21:43.45 | apurvtwr | hariom can you listen to the 16bit audio in audacity? |
| 21:48.44 | AkkerKid | [TK]D-Fender: basically, when I implement this addendum, the calls going through it hit congestion on either line 20 or 46 and error out. |
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| 21:51.11 | [TK]D-Fender | it would probably help to actually look at the call and prove that your configs loaded as well |
| 22:06.55 | hariom | apurvtwr: yea |
| 22:11.52 | apurvtwr | there was a similar but not quite the same problem that I was facing yesterday. It might help if you look at the encodings of both the files. In my case, converting from big-endian to little-endian helped |
| 22:16.45 | hariom | I think tty9 in safe_asterisk enables a terminal tty9. What are the side effects of it if this line is commented? |
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| 22:18.39 | apurvtwr | that's something I am not qualified to answer. ;) |
| 22:26.32 | Preytell | According to this, http://forums.whirlpool.net.au/archive/607081 nothing. |
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| 22:34.46 | hariom | apurvtwr:safe_asterisk opens a console on tty9. This does not require a password and will provide a root shell to anyone passing by. Remove the offending line, or don’t use safe_asterisk |
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| 23:29.57 | thuddwhirr | has anyone ever seen a case where the username of a matching peer is ignored on an incoming INVITE auth? |
| 23:30.44 | ChannelZ | depends on the peer type I suppose |
