| 00:08.48 | ruben23 | p3nguin: goodnews no error anymore just this ---> format_wav.c:201 check_header: Unknown block - not fact or data and file.c:385 fn_wrapper: Unable to open format wav |
| 00:08.53 | ruben23 | any idea for this |
| 00:09.18 | p3nguin | I'm not familiar with that message. |
| 00:37.32 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 00:40.38 | *** join/#asterisk Defraz (~Defraz@69.20.176.132) |
| 00:49.18 | *** join/#asterisk HQuest (~HadjiQues@75.46.34.174) |
| 00:56.01 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-xxmmkhcrdpdfyssj) |
| 00:57.04 | *** join/#asterisk RickCogley_ (~anonymous@ntkngw522154.kngw.nt.ftth.ppp.infoweb.ne.jp) |
| 01:02.11 | HQuest | so, uhh |
| 01:02.51 | HQuest | I'm trying to get an install of AsteriskNow! going as a proof of concept that we can do an in-house digital PBX since we're a small business of only 4-5 people |
| 01:03.14 | HQuest | and one of the conditions is that I have to get the gui up on the server in addition to being able to remote in to it |
| 01:04.05 | HQuest | does any form of X even ship on AsteriskNow!? I tried to install it and even gnome (for my boss's sake) with Yum, but it was a failure to launch |
| 01:04.28 | HQuest | is there a web browser or anything I can just open to see the config gui? |
| 01:10.34 | p3nguin | hquest: Even though this isn't the AsteriskNOW channel, I'll try to explain it briefly... |
| 01:11.07 | HQuest | ohh, sorry. I didn't think to look for a channel specifically for AsteriskNow! |
| 01:11.39 | p3nguin | AsteriskNOW is a CentOS Linux distro with Asterisk pre-installed and your choice of FreePBX, the Asterisk GUI, or no GUI at all. |
| 01:12.27 | p3nguin | If you choose a GUI option, the system is to be administered from the web, from another computer on the network. |
| 01:12.33 | funkylonehat | Hey guys, asterisk 1.8 timing off dahdhi, i have issues with voice quality when people are on calls and supervisors attempt to monitor using chanspy |
| 01:13.28 | funkylonehat | i've checked on the boards, usually a timing issue, but mine's correct. |
| 01:13.29 | p3nguin | If you choose the no GUI option, you administer the system like any regular asterisk deployment -- via ssh or the system console. |
| 01:14.22 | p3nguin | hquest: Does this adequately answer your question? |
| 01:14.26 | *** join/#asterisk corretico (~luis@190.211.94.6) |
| 01:14.37 | corretico | hi |
| 01:14.40 | corretico | I need help |
| 01:15.13 | corretico | When i start asterisk... I get the following message: SETTING FILE PERMISSIONS |
| 01:16.49 | HQuest | p3nguin, not particularly, but I may be better served by the AsteriskNOW or CentOS rooms. What I want to know is how to access the gui locally from the server. I'm not well versed in console commands, and I don't know if any versions of X or any browsers are in the install |
| 01:17.36 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 01:21.23 | p3nguin | hquest: Like I said, if you are using a GUI option, you administer the system from another computer on the network with a web browser. |
| 01:22.14 | *** join/#asterisk snadge (~snadge@unaffiliated/snadge) |
| 01:22.44 | p3nguin | I feel that I addressed that part quite well. |
| 01:22.48 | snadge | i unfortunately have a voip only line.. and occasionally need to send/receive a fax.. like once in a blue moon |
| 01:23.12 | snadge | there are various email -> fax gateways out there.. but the majority of them cost money etc.. and its not something i do often enough to really justify that |
| 01:23.47 | funkylonehat | asterfax? |
| 01:23.50 | snadge | is sending faxes over sip worth bothering with? or is it too problematic |
| 01:24.22 | funkylonehat | depends on how your sip trunk terminates into the PSTN really |
| 01:24.51 | snadge | no idea.. i use "nodephone" which is part of internode, a fairly large aussie isp |
| 01:25.02 | funkylonehat | Similar thing for us, pure sip environment --> cisco gateway --> E1 service. |
| 01:25.03 | HQuest | p3nguin, I understand that, but part of the condition to get this up is that I need to be able to manage it locally right from the server, but I guess I will go to a general linux room or the CentOS room to ask to install it |
| 01:25.23 | funkylonehat | potentially may work, what codec do you use? |
| 01:25.31 | snadge | but it gives me a local phone number.. which i can receive and make calls with |
| 01:25.55 | snadge | well.. i dont actually use it at all to be honest.. and the rare occasion i have, its been with a soft phone |
| 01:26.03 | snadge | i have setup asterisk before for work though, with a similar isp |
| 01:26.18 | p3nguin | hquest: I've given you the answer twice. If you use a GUI option, you manage it from ANOTHER COMPUTER ON YOUR NETWORK. |
| 01:26.20 | funkylonehat | codec? |
| 01:26.57 | snadge | i dont know .. i didnt pay much attention to the codec.. i know one of them costs money and im probably not using that one |
| 01:27.23 | snadge | so i guess considering i just want to be able to send or receive a fax occasionally.. i would use whatever codec is likely to be the most successful for that purpose |
| 01:28.12 | p3nguin | To send faxes, you'll need to use g.711. |
| 01:28.13 | p3nguin | ulaw or alaw |
| 01:28.22 | funkylonehat | yeah. |
| 01:28.28 | snadge | is that proprietary? |
| 01:28.33 | p3nguin | No. |
| 01:28.34 | funkylonehat | no |
| 01:28.37 | funkylonehat | lol |
| 01:28.53 | funkylonehat | provider needs to have fax protocol passthrough enabled |
| 01:28.54 | snadge | ok excellent.. well.. the sucky part is, im using ubuntu.. is that going to be an issue? :P |
| 01:29.02 | p3nguin | No. |
| 01:29.18 | snadge | excellent is packaged for ubuntu |
| 01:30.02 | funkylonehat | I'm having chanspy issues with 1.8 and dahdi timing |
| 01:30.16 | snadge | now i need to read up on asterisk for ubuntu, and asterfax |
| 01:31.18 | p3nguin | I don't know anything about asterfax, but you can use fax for asterisk. |
| 01:31.29 | p3nguin | There's even a free channel license. |
| 01:32.58 | HQuest | are there any free SIP providers around right now? |
| 01:33.04 | HQuest | even just for a single number? |
| 01:33.12 | p3nguin | Yes. |
| 01:33.29 | *** join/#asterisk RickCogley (~anonymous@EM114-48-68-112.pool.e-mobile.ne.jp) |
| 01:33.32 | snadge | which one is easier? fax for asterisk or asterfax? :p |
| 01:33.34 | p3nguin | IPkall will give you a number. |
| 01:33.40 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 01:34.15 | p3nguin | I don't know of any providers offering free termination. |
| 01:35.59 | p3nguin | snadge: FFA is pretty simple to use. Since I don't even know what asterfax is, I can't rate it. |
| 01:49.27 | HQuest | for gui management, do you guys recommend the Asterisk GUI or the FreePBX gui? |
| 01:50.03 | snadge | why is the digium download site asking me what cpu variant im downloading for? |
| 01:50.30 | snadge | barcelona, core2, generic, nocona, opteron and opteron_sse3 |
| 01:50.40 | snadge | i have a bulldozer cpu |
| 01:50.56 | snadge | does it matter? |
| 01:52.03 | *** join/#asterisk DaPrivateer (~matt7229@71-9-155-174.static.oxfr.ma.charter.com) |
| 01:53.39 | p3nguin | Most of us here don't recommend any GUI. GUIs just complicate things beyond all recognition, making it impossible to do things like a normal admin would. |
| 01:54.16 | snadge | i used freepbx at work.. because i couldn't be bothered reading things and learning how they work |
| 01:54.16 | p3nguin | snadge: If you feel like it doesn't matter what kind of CPU you have, get the generic build. |
| 01:54.37 | snadge | but this is apparently ignorant, and lazy.. guilty :p |
| 01:55.15 | p3nguin | I won't touch a system that someone else has crapped up with FreePBX unless it is to remove FreePBX. |
| 01:55.40 | snadge | at the moment im forcing myself to learn about how to make my own email to fax gateway, simply because i can't justify paying a monthly service fee for something i might use twice per year |
| 01:56.09 | snadge | im inclined to go with Fax for Asterisk because its free for a single concurrency and its supported by digium |
| 01:56.16 | p3nguin | If you NEED a GUI, you're probably best to use FreePBX, since there is fairly reasonable support for FreePBX in the #FreePBX channel. |
| 01:57.10 | bbourdage | What is typically the cause of this error ? WARNING[2297]: pbx.c:8134 add_priority: Unable to register extension '174', priority 1 , I am getting them for every extension in the dial plan, I have looked for duplicate includes everywhere ? |
| 01:57.11 | snadge | i think the network latency issues associated with faxing via sip.. may be mitigated somewhat by using the voip provider which is local to the isp im connected to |
| 01:57.23 | snadge | via an adsl2 connection |
| 01:57.39 | snadge | theres one way to find out anyway |
| 01:58.13 | p3nguin | It isn't latency that destroys VoIP -- it's jitter. |
| 01:58.29 | p3nguin | But ridiculously high latency will also be bad. |
| 01:58.48 | snadge | well.. hopefully as the voip server is pretty much local.. the jitter will be better |
| 01:58.56 | snadge | than trying to use a free sip provider on another continent (for example) |
| 01:59.13 | p3nguin | I'd say it's a safe presumption, but you should still check it out to be sure. |
| 02:03.04 | snadge | well i can only try it anyway.. if it works.. yay |
| 02:07.54 | HQuest | thanks for the help, p3nguin |
| 02:09.10 | snadge | hmm.. there doesnt seem to be documentation for using asterisk packages for ubuntu |
| 02:09.19 | snadge | i found reference to installing it from source.. but this appears to be outdated |
| 02:09.42 | snadge | precise (12.04 dev version) comes with asterisk 1.8.4.4 |
| 02:09.46 | p3nguin | It should be in the asterisk wiki. |
| 02:13.31 | p3nguin | If you haven't already found it: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages |
| 02:14.41 | snadge | after all that.. its just apt-get install asterisk .. *facepalm* |
| 02:14.45 | *** join/#asterisk infidel (~crumpe@mail.membersrealm.com) |
| 02:14.58 | snadge | i can apparently use repositories.. but i dunno if thats really necessary at this point |
| 02:16.47 | *** join/#asterisk ketas (~ketas@2001:ad0:91f:0:290:27ff:fea6:a66a) |
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| 02:56.45 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 03:00.36 | *** part/#asterisk ShaunR (~ShaunR@freenode/sponsor/NDChost.com) |
| 03:06.39 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
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| 03:28.15 | dijib | busy night |
| 03:34.03 | snadge | ok so fax for asterisk doesnt mention anything about supporting email |
| 03:34.05 | snadge | or pdf |
| 03:34.35 | snadge | ive installed it.. but i have no idea how to actually use it |
| 03:39.45 | *** part/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 03:40.12 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
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| 03:52.54 | p3nguin | snadge: You'll have to construct that yourself, or at least find where someone else has constructed it and use what they've created. |
| 03:54.41 | p3nguin | snadge: The fax to email part is cake. I can show you how to handle that part within a minute -- it's all dial plan. Email to fax is a little different; you have to set up the interface so that you send email to a specific address and everything gets set it motion. |
| 03:55.32 | p3nguin | If I had to build email to fax, I'd figure out how to run a script when a specific email address receives a mail, then the script would build a call file which would send the fax. |
| 03:58.18 | *** join/#asterisk nzkiwi1 (~michaelnz@ip210-48-102-29.nettrust.net.nz) |
| 03:58.37 | nzkiwi1 | hi |
| 03:58.37 | *** join/#asterisk Defraz (~Defraz@69.20.176.132) |
| 03:59.20 | nzkiwi1 | we have run a lot of tests and we are 98% sure we know what the problem is with yealink T28P and asterisk TLS |
| 03:59.46 | nzkiwi1 | has anyone else had any issues with Yealink and TCP/TLS modes? |
| 04:00.22 | sawgood | You want to use a Yealink with TCP for SIP? |
| 04:00.37 | nzkiwi1 | TLS actually but TCP doesn't work either |
| 04:01.18 | sawgood | oh encryption ... |
| 04:01.37 | sawgood | I have not done that with a Yealink phone, but I've been using Yealink stuff for a long time |
| 04:02.32 | snadge | this is my first time setting up asterisk without freepbx |
| 04:02.50 | p3nguin | Welcome to the real world. |
| 04:02.51 | nzkiwi1 | at this stage I am 98% sure we have nailed it and I just want to find other people wit these problems before going back to Yealink with the results |
| 04:02.53 | snadge | not sure whether to edit the default sip.conf .. or just copy an example |
| 04:03.26 | nzkiwi1 | I code my sip.conf from a blank page |
| 04:03.43 | p3nguin | snadge: The sample files are not suitable for direct drop-in use, but some are very close. |
| 04:03.48 | snadge | well when you see things like "insecure=very" thats a tad concerning |
| 04:04.05 | snadge | this is an example file for my provider nodephone (internode isp) from a few years ago |
| 04:04.12 | snadge | 2008 ;) |
| 04:04.19 | snadge | the other examples are even earlier.. 2005 |
| 04:04.23 | p3nguin | There is no more "very". There's "port", "invite", and "port,invite". |
| 04:04.40 | p3nguin | If they say very, they mean port,invite. |
| 04:05.02 | nzkiwi1 | I think it is only insecure=invitew now |
| 04:05.10 | nzkiwi1 | that works for me |
| 04:05.48 | p3nguin | I just looked over the sample sip.conf, and it looks safe to drop in and modify to your own needs. |
| 04:06.45 | p3nguin | The obsoleted "very" value is equal to "port,invite". insecure=port,invite |
| 04:07.21 | p3nguin | But only use either value if you need it. |
| 04:09.00 | p3nguin | snadge: Do not attempt to use the sample extensions.conf. It is not intended to be used on a real system. |
| 04:10.18 | snadge | first things first i guess.. i just want to be able to place and receive voip calls with my provider |
| 04:10.28 | snadge | then figure out a way to use the installed fax for asterisk.. to send and receive faxes |
| 04:11.51 | p3nguin | Set up a sip peer entry for the provider. Which provider are you using? |
| 04:12.06 | snadge | nodephone from internode (aussie isp0 |
| 04:12.20 | snadge | and i read in their faq that they support t38.. happy days ;) |
| 04:12.25 | p3nguin | Ah, I don't have a peer definition for them. |
| 04:12.49 | snadge | i have found some vague references to peoples config files on the net.. but nothing "official" |
| 04:13.36 | p3nguin | Let me share a couple of my ITSP peer entries. |
| 04:15.16 | p3nguin | Flowroute example: http://pastebin.com/657mLaLm |
| 04:15.57 | snadge | so you dont need a register line? |
| 04:16.36 | p3nguin | One moment. I'll add that. |
| 04:16.44 | snadge | http://forums.whirlpool.net.au/forum-replies.cfm?t=388262&p=2#r21 |
| 04:16.50 | snadge | yay.. that looks helpful :) |
| 04:18.15 | p3nguin | Refresh the flowroute paste. |
| 04:19.06 | p3nguin | VoIP.ms example: http://pastebin.com/fJgNLGLM |
| 04:19.21 | snadge | so [general] is not needed ? |
| 04:19.26 | dijib | root |
| 04:19.53 | p3nguin | Each of these is just an example of a single peer entry. |
| 04:20.01 | p3nguin | You still need the rest of sip.conf. |
| 04:23.54 | p3nguin | Here's a sip.conf example: http://pastebin.com/tER2jGnY |
| 04:24.30 | *** join/#asterisk jamesmills (~jamesmill@2402:c00:2:0:ca2a:14ff:fe3f:6227) |
| 04:25.00 | snadge | ahh cool thanks |
| 04:25.29 | jamesmills | Hey guys. I have this problem I've been trying to work out for the last couple of days. Basically in the pcaps I'm seeing phones sending an INVITE, asterisk not responding, the phone re-sending another INVITE and this occuring 5-6 times with a delay of 5-6s before asterisk finally respond. |
| 04:25.31 | jamesmills | Any ideas? |
| 04:25.53 | jamesmills | Would this have to do with settings in /etc/asterisk/asterisk.conf in [options] ... maxload, maxcalls, etc? |
| 04:26.24 | jamesmills | How do I display what the current settings actually are? The entire [options] section in my asterisk.conf is commented out |
| 04:29.28 | p3nguin | snadge: You may want to refresh that. I had to delete an obsoleted setting. |
| 04:31.18 | snadge | im not sure where the register line is supposed to go |
| 04:32.04 | snadge | inside general.. inside my [nodephone] provider definition |
| 04:32.45 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 04:33.49 | p3nguin | ~book |
| 04:33.49 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 04:33.55 | p3nguin | Have you read the book? |
| 04:34.25 | p3nguin | Register statements must be in the general section, above any peers, and above the authentication section if it exists. |
| 04:35.05 | p3nguin | My sip.conf working example shows where a register statement belongs. |
| 04:38.07 | snadge | yeah your example has it in between alwaysauthreject and localnent |
| 04:38.33 | snadge | its hard when using the example sip.conf as a reference to figure out where you should put it |
| 04:39.07 | p3nguin | The book should explain those details. |
| 04:39.12 | snadge | it might be easier for me to delete all the comments |
| 04:39.19 | snadge | and keep a backup of the original |
| 04:43.05 | p3nguin | As I mentioned, I looked over the included sample sip.conf and it looks like it is safe to drop in and use, but modify it to suit your needs. Use my example as a guideline if necessary. |
| 04:44.27 | *** join/#asterisk darkskiez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 04:48.35 | snadge | cool.. getting there, what is "[00001234FFFF-a] ; phone at exten 123" ? |
| 04:48.44 | snadge | thats an extension right? |
| 04:49.24 | p3nguin | No, that's a phone. |
| 04:49.31 | p3nguin | The phone's name is 00001234FFFF-a |
| 04:49.46 | p3nguin | In that example, the phone named 00001234FFFF-a will be using extension 123. |
| 04:50.47 | p3nguin | Read the commenting in the top of my example which is entitled "Naming devices." |
| 04:51.01 | jamesmills | anyone have any useful ideas for my problem? |
| 04:51.01 | *** join/#asterisk Tyrael1 (~Ryan@c-50-129-214-142.hsd1.in.comcast.net) |
| 04:51.05 | p3nguin | Also... |
| 04:51.09 | p3nguin | ~devices |
| 04:51.09 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
| 04:51.29 | p3nguin | Do not confuse an extension number with a device's name. They are not the same. |
| 04:53.11 | Tyrael1 | Anybody have any Aastra experience? |
| 04:53.38 | snadge | ok well im just going to be using a softphone to connect to it.. with extension 1 |
| 04:53.45 | snadge | which is highly original |
| 04:54.15 | p3nguin | And the phone's device name for going in sip.conf is going to be... ? |
| 04:54.20 | p3nguin | [myphone] ? |
| 04:54.29 | snadge | that will do yeah :) |
| 04:55.07 | p3nguin | So to associate the phone with extension 1, in the dial plan, you'll end up with: exten => 1,1,Dial(SIP/myphone) |
| 04:55.27 | snadge | ok that makes sense |
| 04:55.50 | p3nguin | In my example, you can see how I made an association using the callerid and the accountcode. |
| 04:57.24 | snadge | ahh i see you didnt paste an example extensions.conf to go with it |
| 04:58.29 | *** join/#asterisk RickCogley (~anonymous@114.179.8.185) |
| 04:58.37 | p3nguin | Just a minute and I will. |
| 05:04.43 | *** join/#asterisk freetown2 (~freetown2@ipvpn098241.netvigator.com) |
| 05:08.44 | p3nguin | Here's my example extensions.conf: http://pastebin.com/Piqv4Egj |
| 05:12.02 | Tyrael1 | anybody have any aastra experience, specifically with using a softkey to park. Preferably not as a speedial that hits #700 ? |
| 05:14.31 | p3nguin | snadge: That example extensions.conf also has the fax-to-email dial plan in it. |
| 05:14.32 | snadge | cheers.. i just kind of realised that you cant run asterisk and a softphone on the same box |
| 05:14.35 | ChannelZ | I don't think parking is a function like hold.. it's only a feature code (?) |
| 05:14.36 | *** join/#asterisk abesamthomas (~abesamtho@115.111.177.122) |
| 05:14.51 | snadge | its complaining about port 5060 being in use ;) |
| 05:14.53 | p3nguin | You /can/ run them both on the same system, but don't. |
| 05:15.18 | [TK]D-Fender | You can run them on the same system... just change the PORT on one |
| 05:15.19 | p3nguin | Depending on the phone, you may be able to change the server port of the phone. |
| 05:15.54 | snadge | yeah its twinkle.. it has a SIP port option which defaults to 5060 and an RTP option which defaults to 8000 |
| 05:16.07 | p3nguin | It is easy to change in twinkle. |
| 05:16.16 | p3nguin | Just change it to something like 5061. |
| 05:16.18 | Tyrael1 | @ChannelZ I think you're right, I have just encountered documentation that said something about putting asterisk:700 in the park button's config... My boss insists it can be done without DTMF tones... |
| 05:17.42 | ChannelZ | well I should say you can transfer to a park extension, it doesn't have to actually be a feature-code #700 type thing |
| 05:18.00 | ChannelZ | so if you can macro that in the phone somehow.. |
| 05:18.32 | Tyrael1 | hmmm... sounds like it could just work... give me a few to try |
| 05:20.32 | Tyrael1 | only thing it does that i dont like then is that it will announce to the caller what park they are... but I can turn that off right? then just rely on BLFs |
| 05:20.48 | Tyrael1 | but you're right... setting it up as a transfer does work |
| 05:22.09 | snadge | hmm.. registration failed.. no matching peer found |
| 05:22.24 | p3nguin | I guess you put the wrong thing in the fields. |
| 05:22.25 | ChannelZ | I've never actually used feature code park, but transfer park should only announce to the person parking, not the parked. |
| 05:22.40 | p3nguin | What did you name your phone in sip.conf? It's the name between the square brackets. |
| 05:24.30 | snadge | myphone |
| 05:26.05 | p3nguin | In the User profile, you put an arbitrary name for Your name, myphone in User name and in Authentication name, and whatever secret you defined in the Password field? |
| 05:26.26 | snadge | ahh thats the problem |
| 05:26.48 | snadge | my name is set to "myphone" username is set to "1" |
| 05:27.22 | p3nguin | You indicated that 1 would be the extension number used to reach the phone. The phone does not need to know the extension number. |
| 05:28.44 | snadge | i made an assumption that accountcode=1 would set the username to 1 |
| 05:29.38 | *** join/#asterisk funkylonehat (~funkylone@125-236-222-73.adsl.xtra.co.nz) |
| 05:31.47 | p3nguin | That's for accounting purposes. |
| 05:35.55 | snadge | excellent.. i now get a guy with an aussie accent saying "the number you have dialled is not in service, please check the number before dialing again" :p |
| 05:36.07 | snadge | so i know thats not an asterisk error.. thats coming from the provider |
| 05:37.05 | *** join/#asterisk ayrjola (~ayrjola@89.18.236.11) |
| 05:39.16 | p3nguin | You also configured a peer for your ITSP? |
| 05:39.26 | p3nguin | And made relevant dial plan to support it? |
| 05:40.18 | snadge | kind of.. im working from |
| 05:40.28 | snadge | http://forums.whirlpool.net.au/forum-replies.cfm?t=388262&p=2 |
| 05:40.53 | snadge | except i removed the leading 0 from the front.. i dont want to dial 0 to access outside line |
| 05:42.31 | snadge | i see what i've done |
| 05:42.41 | snadge | it should just be $EXTEN |
| 05:43.34 | snadge | err without the :1 i mean |
| 05:50.35 | snadge | i cant believe i have gotten this far.. you are very patient p3nguin .. sorry about that :| |
| 05:56.53 | snadge | [from-nodephone] |
| 05:56.54 | snadge | ; Ring the internal SIP handsets: |
| 05:56.56 | snadge | exten => s,1,Dial(SIP/myphone) |
| 05:57.28 | snadge | i get the person is unavailable message when i dial in.. so that line there is obviously wrong and even though it should be, its not obvious to me why |
| 05:58.10 | p3nguin | You've set the context for your ITSP to be context=from-nodephone? |
| 05:58.47 | snadge | i think so.. if itsp is where i've got [nodephone] |
| 05:59.58 | p3nguin | You'll need to determine if they are sending calls to extension 's' or to an extension which is your phone number. |
| 06:02.02 | snadge | i just had a look with sip set debug on |
| 06:03.25 | snadge | <--- SIP read from UDP:203.2.134.1:5060 ---> |
| 06:03.25 | p3nguin | It will say "Looking for ... in 'from-nodephone'" |
| 06:03.33 | snadge | INVITE sip:s@192.168.1.10:5060 SIP/2.0 |
| 06:09.39 | *** join/#asterisk cyborg-one (1000@79.140.12.17) |
| 06:09.50 | snadge | only clue i can find is this |
| 06:09.52 | snadge | Using INVITE request as basis request - BW1638558432202121164306736@203.2.134.129 |
| 06:09.52 | snadge | Found peer 'nodephone' for '0447308627' from 203.2.134.1:5060 |
| 06:09.52 | snadge | <--- Reliably Transmitting (NAT) to 203.2.134.1:5060 ---> |
| 06:09.56 | snadge | SIP/2.0 401 Unauthorized |
| 06:13.55 | snadge | possibly need one of those insecure options |
| 06:15.17 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 06:16.46 | *** join/#asterisk FainaUkraina (~Gene@203.145.92.210) |
| 06:16.57 | snadge | insecure=port,invite worked.. :P |
| 06:20.03 | *** join/#asterisk Srini (Srinivasa@2002:74cb:81f6::74cb:81f6) |
| 06:20.45 | Srini | If I were to define carriers for TE220, the protocal is ZAP? |
| 06:21.19 | WIMPy | Zap is dead. And what does "define carrier" mean? |
| 06:29.30 | Srini | WIMPy, I know it is not correct to take vicidial here, but in the carrier definition of vicidial it says Carrier Protocal Zap/SIP/External |
| 06:33.00 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 06:38.24 | WIMPy | Not updated in the last 4 years? |
| 06:39.44 | kaldemar | Srini: translated into asterisk terminology, you mean technology. |
| 06:40.45 | *** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 06:41.14 | kaldemar | Srini: if your system uses zaptel, then it's "Zap". zaptel was renamed to DAHDI years ago, so if your system uses DAHDI, the technology part is "DAHDI". |
| 06:42.45 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-owpaevhvzggufvdv) |
| 07:03.33 | *** part/#asterisk jamesmills (~jamesmill@2402:c00:2:0:ca2a:14ff:fe3f:6227) |
| 07:04.34 | ayrjola | Can I set to uri to be different from request uri, if yes how..? Trying to duplicate setup with some stupid PBX that sends request uri with + prefix and to uri without + prefix. |
| 07:04.42 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:05.09 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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| 07:38.05 | snadge | how do i redirect from-nodephone to fax-in ? |
| 07:38.07 | snadge | currently i have |
| 07:38.14 | snadge | exten => s,1,Dial(SIP/myphone) |
| 07:38.42 | snadge | which i used redirect to the incoming call to a softphone |
| 07:39.42 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:39.43 | schmidts | good morning |
| 07:41.05 | snadge | i think i got it.. change Dial to Goto(fax-in,fax,1) |
| 07:44.49 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 07:44.58 | snadge | ${DB(fax/fax-manager/email)} |
| 07:45.01 | snadge | where do i set that variable ? |
| 07:47.48 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 07:49.55 | *** join/#asterisk BuenGenio (~Gene@203.145.92.210) |
| 07:52.41 | wdoekes2 | morning |
| 07:53.16 | krotos | morning too |
| 07:55.38 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 08:02.14 | *** join/#asterisk DarthExpeditor (~IceChat9@96-42-133-130.static.trcy.mi.charter.com) |
| 08:02.57 | kaldemar | snadge: depends entirely on what you want to do with it. and it's not a variable, but a function call which reads a value (family fax, key fax-managet/email) from asterisk's database. |
| 08:07.12 | schmidts | snadge as far as i know you can use set for it like Set(DB(fax/fax-manager/email)=test@test.com) |
| 08:09.31 | snadge | im so close to getting fax for asterisk working.. with limited asterisk knowledge, i knew it wouldn't be easy |
| 08:10.06 | schmidts | snadge its not so hard as you think ;) |
| 08:10.07 | snadge | its not like im trying to make money out of it.. i just want to be able to send/receive faxes.. the annoying part is, i have a naked adsl2 line.. and a physical fax machine.. well an mfd |
| 08:10.45 | snadge | and for the 3 or 4 faxes i send/receive in a year.. i dont want to shell out a monthly subscription for a fax2email gateway ;) |
| 08:12.37 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
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| 09:00.27 | *** join/#asterisk Dovid (~Dovid@office.mypbxmanager.net) |
| 09:00.52 | Dovid | if i have directrtpsetup=yes why does asterisk first send it's own IP and then request a re-invite? |
| 09:01.21 | snadge | oh my god.. i just received a fax via sip.. *successkid* ;) |
| 09:01.25 | snadge | welcome to the 80s |
| 09:01.45 | WIMPy | It's all about the 80s here. |
| 09:02.00 | snadge | im a bit disappointed that 14.4k is the limit :P |
| 09:02.12 | snadge | http://en.wikipedia.org/wiki/Fax |
| 09:02.23 | snadge | according to wikipedia there is a 33.6k fax standard |
| 09:02.30 | snadge | and apparently there are colour faxes too |
| 09:02.35 | snadge | not mentioned in that article |
| 09:03.04 | WIMPy | And there's even 64k fax. |
| 09:03.05 | kaldemar | Dovid: because you have directrtpsetup=yes. :P that's the parameter that enables that functionality. the new invites are to set up the RTP directly between the devices and leave asterisk out of the path. |
| 09:04.16 | Dovid | kaldemar: How do I do it that the initial invite as the the others IP so there is no need for a re-invite? |
| 09:04.26 | snadge | has someone got a fax number i can test my outbound fax with? |
| 09:04.59 | snadge | i just want to verify that it works before i start faxing things off all over the place ;) |
| 09:05.03 | Dovid | snadge: What country? |
| 09:05.31 | snadge | im in australia.. but i think my provider has reasonable rates to the usa/canada etc.. it would just be a single page anyway |
| 09:05.48 | wdoekes2 | on pstn I do, in NL |
| 09:06.19 | Dovid | testung untl. is never really testing since many carriers have issues |
| 09:06.25 | snadge | netherlands? |
| 09:06.32 | *** join/#asterisk Nasga (~Nasga@82.113.117.78.rev.sfr.net) |
| 09:07.10 | wdoekes2 | correct |
| 09:07.45 | snadge | 5c/min |
| 09:07.55 | snadge | luls thats cheaper than a local mobile |
| 09:08.21 | snadge | hmm international space station is $10/min |
| 09:08.50 | WIMPy | o.O Are they their own country? |
| 09:09.08 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 09:09.20 | Dovid | How do I do it that the initial invite as the the others IP so there is no need for a re-invite? |
| 09:09.37 | snadge | 29c/min to a mobile in my own country |
| 09:10.50 | snadge | ok the next question is.. how do you send a fax without using a channel originate command |
| 09:11.29 | wdoekes2 | using ami or a callfile |
| 09:12.18 | kaldemar | Dovid: ahem, i misread your initial parameter. directrtpsetup is supposed to do that. are the devices behind a NAT of configured in sip.conf to be? |
| 09:12.49 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 09:13.38 | kaldemar | Dovid: directmedia or the nat setting may need to be disabled for directrtpsetup to work. |
| 09:15.01 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 09:20.48 | snadge | im pretty sure just about everyones thought of converting goatse into a tiff file and faxing it |
| 09:21.30 | molnarp | hi, i'm looking for someone familiar with, or access to a Digium TDM410P card with VPMADT032 echo canceller module |
| 09:28.38 | Chainsaw | molnarp: I have a TDM410P, but no echo cancellation hardware plugged into it. Sorry. |
| 09:28.48 | Chainsaw | molnarp: (I use it as a fax board, so no point) |
| 09:29.42 | molnarp | Chainsaw: I see, thanks anyway. I have issues regarding firmware loading to the VPMADT032 |
| 09:29.44 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 09:30.01 | Chainsaw | molnarp: You have the power plugged in? |
| 09:30.21 | Chainsaw | molnarp: You can get away without it with FXO modules only, but FXS modules and the echo cancellation need power last I checked. |
| 09:30.21 | WIMPy | What issue? |
| 09:30.35 | molnarp | I only have FXO modules, AFAIK no power neccessary in this case |
| 09:30.57 | Chainsaw | molnarp: I'm fairly sure that the echo cancellation depends on external power being connected. |
| 09:31.07 | molnarp | are you sure about that the echo canceller needs external power? |
| 09:31.18 | molnarp | well, i'm gonna try this |
| 09:31.20 | molnarp | thanks |
| 09:31.26 | Chainsaw | molnarp: Not enough to testify in an open court, no. But consider it an educated guess. |
| 09:32.17 | molnarp | WIMPy: the issue is that the echo canceller module doesn't get regognized |
| 09:32.59 | WIMPy | Not recognizeed or not activated? |
| 09:33.30 | molnarp | not recognized at all, the wctdm24xxp prints VPM100: Not installed |
| 09:33.47 | WIMPy | Have you tried to load themodule with debug and look at dmesg? |
| 09:34.07 | *** join/#asterisk BuenGenio (~Gene@203.145.92.210) |
| 09:34.16 | molnarp | i'd like to know, what does the wctdm24xxp module prints in case the module is present, but no firmware/wrong firmware is loaded? |
| 09:34.50 | molnarp | i looked at dmesg, but not with debug |
| 09:35.02 | molnarp | how do i load the module with debug? |
| 09:35.22 | WIMPy | modprobe ... debug=255 or something. |
| 09:35.37 | molnarp | trying |
| 09:35.45 | WIMPy | I have only tried it with a pri card. |
| 09:36.21 | WIMPy | That gave a lot of debug, but still rather little about the VPM. |
| 09:36.44 | *** join/#asterisk stephendowed (~stephen@197.255.215.252) |
| 09:37.29 | stephendowed | so am new to asterisk and i run it on a vm on my system, my systm has a public address and so does the vm since i bridged it to my wireless to also get a public address |
| 09:37.30 | molnarp | hmm |
| 09:37.31 | Dovid | kaldemar: i stepped a way for a bit |
| 09:37.32 | molnarp | <PROTECTED> |
| 09:37.36 | stephendowed | problem is |
| 09:37.38 | molnarp | Failed: Sent 0 != ff VPMADT032 Failed HI page test |
| 09:37.45 | Dovid | kaldemar: thats exactly what i have set up. i will pb my config |
| 09:38.18 | stephendowed | when i try to call across two soft phones one on another system with a public id and one on the system i run the vm on it does not work |
| 09:38.28 | WIMPy | molnarp: So it fails before any firmware loading? |
| 09:38.32 | stephendowed | form my system i can call the other softphone on another computer |
| 09:38.40 | Dovid | kaldemar: thats exactly what i have set up. i will pb my confi |
| 09:38.45 | Dovid | kaldemar: http://pastebin.com/3E62A7s6 |
| 09:38.54 | molnarp | apparently |
| 09:38.57 | stephendowed | the other i cant is shows me something abt forwarding it to channel_local wich does not exist |
| 09:39.13 | molnarp | it's important to mention that is use debian stable (squeeze) |
| 09:39.26 | molnarp | therefore my packages are quite old |
| 09:39.38 | WIMPy | molnarp: If that's important, that's bad. |
| 09:39.43 | WIMPy | That one. |
| 09:40.09 | WIMPy | Anyway: If it fails that early that looks like a hardware issue. |
| 09:40.19 | WIMPy | Did you give power to it? |
| 09:41.14 | molnarp | nope, as far as I know external power is only required for FXS modules |
| 09:41.21 | molnarp | i only have an FXO module |
| 09:41.33 | WIMPy | It's worth a try. |
| 09:41.50 | WIMPy | Or try to remove and reattech the module. |
| 09:42.16 | molnarp | apparently, i'm gonna try it, next time i have physical access to that box |
| 09:42.34 | molnarp | thank you for your help |
| 09:43.43 | stephendowed | so anyone has an idea? please help a newbie |
| 09:45.29 | kaldemar | Dovid: directmedia is yes by default, you need to set it to "no" to disable it, not just comment out the yes setting. addition to that, you have canreinvite=yes. careinvite was renamed to directmedia but it still works as another name for directmedia. |
| 09:46.23 | molnarp | WIMPY i tried to remove and reattach yesterday |
| 09:47.31 | Dovid | kaldemar: so I need directmedia=no and directrtpmeida=yes ? |
| 09:48.06 | kaldemar | Dovid: directrtpsetup, not directrtpmedia |
| 09:49.26 | stephendowed | no answer to ma question? (sad face) |
| 09:50.36 | Dovid | kaldemar: http://pastebin.com/pN6VkCxh |
| 09:50.36 | Dovid | ? |
| 09:50.45 | Dovid | that has asterisk holding on to the media |
| 09:51.43 | kaldemar | Dovid: what do you see in sip debug with those settings and with directmedia enabled? |
| 09:52.15 | Dovid | kaldemar: with the ones I just pb'd? invite out has asterisk ip for rtp |
| 09:52.41 | *** join/#asterisk BuenGenio (~Gene@203.145.92.210) |
| 09:54.49 | kaldemar | Dovid: in both cases? maybe you should pastebin. |
| 09:55.05 | Dovid | kaldemar: It alaw |
| 09:55.29 | Dovid | kaldemar: well if i have both set to yes and nat=no then it will use it's own ip and then after 200 OK it does re-invite |
| 09:55.37 | Dovid | i dont want re-invite. i want direct rtp |
| 10:03.05 | *** join/#asterisk bitwize (~bitwize@h87-96-213-2.dynamic.se.alltele.net) |
| 10:03.18 | ayrjola | Can I set to uri to be different from request uri, if yes how..? Trying to duplicate setup with some stupid PBX that sends request uri with + prefix and to uri without + prefix. |
| 10:11.38 | *** join/#asterisk ccesario (~ccesario@187.17.166.162) |
| 10:19.33 | stephendowed | please how do i make a client behind a nat register to asterisk and make calls |
| 10:21.46 | stephendowed | any one ther? |
| 10:22.35 | kaldemar | ~sipnat |
| 10:22.36 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 10:25.10 | stephendowed | what does the canreinvite thing to in plain terms please |
| 10:26.33 | schmidts | stephendowed canreinvite or in newer asterisk version directmedia makes asterisk tries to bridge sip peers directly together so the audio stream from one phone to another will not go through asterisk but you will notice some problems if one client is behind nat |
| 10:27.17 | stephendowed | alright so in a case one of them is behind a nat and d other has a public ip as well as asterisk what can i do to make dem talk |
| 10:27.39 | stephendowed | cos even the one behind d nat refuses to register @all |
| 10:32.27 | *** join/#asterisk atekinalp (~alper@85.105.28.77) |
| 10:35.13 | kaldemar | stephendowed: what do you mean by refuses to register? does your asterisk get a registration message? |
| 10:35.13 | stephendowed | alright so in a case one of them is behind a nat and d other has a public ip as well as asterisk what can i do to make dem talk |
| 10:35.13 | stephendowed | <stephendowed> cos even the one behind d nat refuses to register @all |
| 10:35.41 | kaldemar | those links have all the information you need. |
| 10:35.52 | stephendowed | @kaldemar no it does not and am using a softphone and when i issue pings from the computer it can reach the asterisk server |
| 10:37.53 | kaldemar | stephendowed: it does not matter what kind of a phone you use. being able to ping is irrelevant also, SIP messages don't run on top of ICMP. |
| 10:38.42 | kaldemar | stephendowed: enable sip debug in asterisk's CLI with "sip set debug on" and make the phone try to register. do you see any messages from the phone in the sip debug? |
| 10:39.08 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 10:39.26 | stephendowed | @kaldemar i will do that but then i have another problem |
| 10:39.57 | kaldemar | let's worry about the other problem later. do you see SIP messages from the phone in asterisk? |
| 10:40.00 | stephendowed | i also ran asterisk atop a vm and bridged it to ma wireless card so it gets a public ip too...phones on ma local system registers |
| 10:40.30 | stephendowed | but other registerd phones cannot call it it shows something abt forwarding the calls to a local channel or so |
| 10:41.16 | kaldemar | from your last two comments i understand that all your phones register successfully. is that correct? |
| 10:50.41 | stephendowed | @kaldemar yes dey did.sorry dis is coming late i had to take care of something |
| 10:52.15 | kaldemar | stephendowed: then pastebin your sip.conf (mask secrets), extensions.conf and a CLI output of a call with sip debug enabled so someone can tell you what is wrong. |
| 11:03.21 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:06.05 | *** join/#asterisk dandate2 (~dan@180.190.222.190) |
| 11:06.38 | dandate2 | so i'm thinking about buying some digital amplifiers but the only deals i can find are for EU standard models, will these be compatible with my cisco 7940 ip phones? |
| 11:11.34 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 11:21.00 | *** join/#asterisk bitwize (~bitwize@h87-96-213-2.dynamic.se.alltele.net) |
| 11:24.16 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 11:24.20 | v0lZy | hi |
| 11:24.51 | v0lZy | Mr. WIMPy, are you here? |
| 11:25.18 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 11:25.36 | *** part/#asterisk jaxon007_ (~jayesh@123.252.144.92) |
| 11:29.03 | *** join/#asterisk Netgeeks (~chris@173.11.68.156) |
| 11:30.13 | saxa | hello, anybody in here willing to help me understand why I do not see a caller id in the CALLERID variable , but I see it in the DEBUG message ? |
| 11:30.20 | saxa | see here: http://pastebin.com/faFVBt2c |
| 11:31.29 | kaldemar | saxa: CALLERID(all) => ${CALLERID(all)} |
| 11:36.08 | saxa | hi kaldemar ok let me see what I have in extensions.conf |
| 11:37.31 | saxa | of course :) |
| 11:37.38 | *** join/#asterisk srini (~Srinivasa@219.91.145.198) |
| 11:37.44 | srini | Hi room |
| 11:38.08 | srini | in a dialplan when I say SIP/${EXTEN} does it mean any availalbe sip extension? |
| 11:38.42 | saxa | I don't know since i had it before with NoOp(${CALLERID(all)}) but then I changed it to Verbose(CALLERID(all)) for some reason :) I think I saw it somewhere on the net. |
| 11:38.42 | kaldemar | srini: no. EXTEN is a variable that holds the current extension in the dialplan. |
| 11:39.28 | kaldemar | saxa: variables and functions need to be surrounded by ${} to be referenced to a value. |
| 11:39.32 | srini | kaldemar, What should I specify for "Any available SIP extension"? |
| 11:39.50 | kaldemar | srini: there is no such thing. |
| 11:40.14 | kaldemar | srini: also, by extension you mean a device. extensions are parts of dialplan in asterisk,. |
| 11:40.31 | WIMPy | hi v0lZy |
| 11:41.44 | srini | So, the that case we have to have list of phones to call in a variable, and then that variable can be called in the dialplan in order to have range of phones which can be called? |
| 11:42.14 | saxa | kaldemar: yes, of course. Thanks anyway , i have not even notices that. |
| 11:42.59 | saxa | srini: you can always use some kind of range of extensions |
| 11:43.02 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 11:43.04 | v0lZy | Hi WIMPy , hi kaldemar |
| 11:43.14 | kaldemar | srini: or list the devices in the dial command, like Dial(SIP/foo&SIP/bar) |
| 11:43.42 | v0lZy | WIMPy: reality check... my ISP just routes a sip trunk to me... no username, no password.... they claim their configuration doesnt support username authentication |
| 11:43.45 | srini | kaldemar, so we cannot generalize them? |
| 11:44.12 | v0lZy | is this kind of bullshit or is there any truth to this? I cant understand why you'd hard route traffic that way... |
| 11:44.13 | kaldemar | srini: but what are you really aiming at? dialing multiple phones at once or one single phone from a defined group of phones? |
| 11:44.56 | srini | Dialing sigle phone from a defined group depending on which one is free |
| 11:45.15 | WIMPy | v0lZy: As we don;t know their software it's hard to say. Maybe they just don't know how to use it, or did a config system that doesn't support it. |
| 11:45.32 | kaldemar | srini: maybe a queue is what you want. see "core show application Queue" and queues.conf.sample. |
| 11:45.54 | kaldemar | srini: or maybe http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-ACD.html would be a better start. |
| 11:46.11 | v0lZy | might be that they dont know how to use it a |
| 11:46.31 | v0lZy | I mean... seriously... its friggin stupid they want me to do 1 to 1 nat for rtp.... |
| 11:47.01 | v0lZy | funny thing, got it working yesterday |
| 11:47.03 | WIMPy | I don't see what that has to do with RTP. |
| 11:47.03 | v0lZy | but its not working now |
| 11:47.06 | srini | kaldemar, Great Help! Thanks! Looks like thats what I am looking for! |
| 11:52.16 | v0lZy | alwso |
| 11:52.29 | v0lZy | WIMPy: i'm getting SIP from one ip (signalisation) they call it) |
| 11:52.32 | v0lZy | and RTP from another |
| 11:55.33 | snadge | im so close.. i have t38 incoming faxes working.. but i cant send |
| 11:56.49 | v0lZy | WIMPy: ... huh.... Feb 22 12:51:01 asterisk[1462]: NOTICE[1497]: chan_sip.c:25290 in sip_poke_noanswer: Peer 'SIP-PROVIDER-19089295184f44cc87c3de5' is now UNREACHABLE! Last qualify: 0 |
| 11:57.16 | snadge | i have a script which generates a call file.. but it doesnt appear to work, im not sure how to correctly invoke it |
| 11:57.35 | v0lZy | whats this sip_poke_noanswer |
| 11:58.11 | kaldemar | snadge: make sure your script does not copy but moves the file to the spool. it needs to be an atomic filesystem operation. |
| 11:58.34 | WIMPy | v0lZy: You have qualify enabled. |
| 11:58.47 | v0lZy | whats qualify? |
| 11:59.11 | stephendowed | @wimpy so the qualify just helps to show if an extension is reachable right? |
| 11:59.13 | WIMPy | Connectivity check. |
| 11:59.27 | stephendowed | if it goes down or offline that also helps us track it right? @wimpy |
| 11:59.34 | WIMPy | peer, not extension |
| 11:59.41 | WIMPy | yes |
| 11:59.54 | v0lZy | so basically it checks every 2 seconds |
| 11:59.59 | v0lZy | and its sayingthat its unreachable |
| 12:00.01 | v0lZy | ? |
| 12:00.12 | WIMPy | And if it has already been qualified as lagged or offline, Dial() won;t even try to reach it. |
| 12:00.36 | WIMPy | Every 2s seems very often. |
| 12:01.47 | v0lZy | btw |
| 12:01.51 | v0lZy | if i dont have a username and password |
| 12:02.10 | v0lZy | i probably dont need to have fromuser field in sip.conf right? |
| 12:02.27 | v0lZy | increased to 60s |
| 12:02.32 | stephendowed | @wimpy is there a way of increasing the frequency it checks with |
| 12:02.55 | v0lZy | i increased it to 60s |
| 12:04.00 | WIMPy | v0lZy: I don;t know what your provider expects. |
| 12:04.50 | v0lZy | ok |
| 12:04.51 | WIMPy | stephendowed: qualifyfreq |
| 12:04.51 | v0lZy | seems to work |
| 12:08.13 | stephendowed | and that is in seconds right |
| 12:08.28 | stephendowed | like qualifyfreq = 5 sets it to every 5s right? |
| 12:11.41 | v0lZy | ok |
| 12:11.44 | v0lZy | I configured my isp |
| 12:11.48 | v0lZy | the qualify thing seems to have done it |
| 12:11.51 | v0lZy | WIMPy: one more question |
| 12:11.56 | v0lZy | in my current situation |
| 12:12.13 | v0lZy | i can call into my phone using a public phone number |
| 12:12.18 | v0lZy | my phone rings |
| 12:12.25 | v0lZy | but i dont hear anything |
| 12:12.29 | v0lZy | and neither does the other side |
| 12:12.58 | v0lZy | what would this imply? |
| 12:14.55 | snadge | the script im using to send faxes is on this link: http://www.teamforrest.com/blog/156/integrating-fax-for-asterisk/ |
| 12:14.58 | snadge | and called faxout.pl |
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| 12:16.07 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:16.13 | snadge | ive verified its generating the pdf and the call file correctly.. its just im not sure exactly how to envoke it.. i did a sudo -u asterisk -p asterisk faxout.pl faxnum /tmp/pdffile.pdf |
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| 12:20.30 | kaldemar | snadge: that generated call file assumes that you have a context called outboundialcontext which has an extension that matches faxnum. once that channel answers, the other end of the call is connected to extension s in a context called outboundfax. |
| 12:22.56 | snadge | yes.. i have changed outboundialcontext to nodephone-faxout |
| 12:23.12 | snadge | i also have an extension s and outboundfax |
| 12:23.30 | v0lZy | im a bit lost |
| 12:23.34 | v0lZy | i get the ringing and all |
| 12:23.39 | v0lZy | but no sound gets transmitted. |
| 12:25.00 | snadge | ahh i think that could be an issue.. i dont think it has an extension that matches faxnum |
| 12:25.32 | snadge | [nodephone-faxout] |
| 12:25.33 | snadge | exten => _X.,1,Dial(SIP/*38#${EXTEN}@nodephone) |
| 12:26.39 | snadge | unless _X. catches it .. the *38# prefix apparently enables t38 support on outbound calls |
| 12:27.21 | kaldemar | snadge: that matches anything that starts with a digit and has length greater than one. |
| 12:28.05 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 12:28.48 | snadge | yeah im purely trying to get fax send/receive working at the moment.. i dont use this voip number for anything else |
| 12:28.55 | snadge | at least receive works :) |
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| 12:31.28 | kaldemar | snadge: what do you see in CLI when you invoke the script? |
| 12:32.20 | snadge | that its hungup after 6400 ms |
| 12:33.02 | kaldemar | what else? "core set verbose 10" |
| 12:33.17 | snadge | it could be a wrong number.. or that i cant send faxes to the netherlands ;) |
| 12:33.27 | snadge | from australia |
| 12:33.35 | snadge | maybe i need a better test number |
| 12:36.40 | mtbf | Hey guys, I have to put a limit of calls per number each day, I have an AGI script, which runs SQL query on CDR table, so I can determine calls count per current day, but as I noticed, records are added into CDR table when the call is finished, therefore making a call which has not been finished yet (and is currently on hold, for instance) invisible to another instance of that AGI script, so the limit can be bypassed. Any suggestions how to solve this one? |
| 12:37.26 | kaldemar | mtbf: don't use CDR. |
| 12:37.59 | mtbf | So tell me please, what to use instead. |
| 12:39.09 | mtbf | I could deploy my own db table for this purpose, but I think this would be redundant and it has to be feasible by some more elegant way. |
| 12:39.56 | kaldemar | mtbf: something custom that you increment from dialplan every time when there is a call. |
| 12:40.45 | kaldemar | mtbf: are you only counting answered calls? |
| 12:41.18 | mtbf | Yes, I'm skipping those with 'BUSY' and 'FAILED' disposition. |
| 12:42.38 | *** join/#asterisk RickCogley_ (~anonymous@114.179.8.185) |
| 12:43.01 | kaldemar | you could check DIALSTATUS after calls and revert the incrementation in non-answered cases. |
| 12:44.16 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 12:44.29 | mtbf | Ok, thnx for your input. |
| 12:44.31 | kaldemar | mtbf: or make the incrementation in a subroutine that is called from the Dial app with option U(). it is executed for the called channel but you can pass info as arguments from the calling channel. |
| 12:44.47 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 12:44.49 | kaldemar | mtbf: that subrouting is executed upon an answer only. |
| 12:44.51 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 12:44.52 | v0lZy | back |
| 12:44.54 | v0lZy | hmmm |
| 12:44.59 | v0lZy | WIMPy: I half solved my problem |
| 12:45.06 | v0lZy | Now i can send sound out |
| 12:45.12 | v0lZy | so people callling my number can hear me |
| 12:45.15 | v0lZy | but i cant hear them |
| 12:45.20 | v0lZy | this appears to be a firewall issue... |
| 12:50.44 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 12:54.58 | snadge | UDPTL asked to send 59 bytes of IFP when far end only prepared to accept 30 bytes; data loss will occur.You may need to override the T38FaxMaxDatagram value for this endpoint in the channel driver configuration. |
| 12:54.59 | snadge | hmm |
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| 13:08.19 | treborsux | I have a problem |
| 13:10.00 | treborsux | Randomly when a user picks up a call whether it be from outside dahdi or internal extension the user just hears ringing like they made a call. If they hit hold and resume the ringing goes away and they continue the call. |
| 13:11.29 | *** join/#asterisk srini (~Srinivasa@219.91.145.198) |
| 13:13.00 | srini | In a dial plan I am using dial(DAHDI/1/${EXTEN}) it is working properly, both incoming and outgoing are ok... what will be span two on that card DAHDI/2? |
| 13:14.29 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:14.46 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 13:20.26 | kaldemar | srini: neither of those mean span, but a channel. there are only individual channels and channel groups that can be selected. |
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| 13:21.54 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:25.32 | v0lZy | im totally confused |
| 13:25.34 | v0lZy | guys |
| 13:25.41 | v0lZy | i can send sound out |
| 13:25.44 | v0lZy | but i cant get sound in. |
| 13:25.51 | v0lZy | i can get ring in |
| 13:25.54 | v0lZy | but not sound. |
| 13:26.03 | v0lZy | ring seems to depend on 5060 being open |
| 13:26.04 | *** join/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
| 13:26.13 | v0lZy | and sound would suggest that it depends on RTP.... |
| 13:26.23 | v0lZy | for which i have 1:1 nat set |
| 13:26.28 | v0lZy | they hear me, i cant hear them |
| 13:27.29 | [TK]D-Fender | ~sipnat |
| 13:27.29 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 13:27.31 | [TK]D-Fender | ^^^ |
| 13:27.46 | v0lZy | but having 1.1 nat.... |
| 13:27.53 | v0lZy | probably means my pbx is the same as if on public ip... |
| 13:28.10 | phpboy | eish, SIP and NAT *shudder* |
| 13:28.31 | v0lZy | my pbx can handle that fine though |
| 13:28.51 | v0lZy | or not.. |
| 13:29.02 | phpboy | if it can handle it fine then what's the problem? |
| 13:29.13 | v0lZy | i dont know |
| 13:29.19 | v0lZy | i mean... this is my setup |
| 13:29.28 | v0lZy | <pbx>---<router>---<provider> |
| 13:29.40 | v0lZy | and i can dial in just fine |
| 13:29.50 | phpboy | what is not happening that's supposed to happen? |
| 13:29.51 | [TK]D-Fender | v0lZy, show us the SIP DEBUG for your failed call & verbose 10 |
| 13:29.53 | [TK]D-Fender | ~pb |
| 13:29.53 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 13:29.54 | [TK]D-Fender | ^^^ |
| 13:29.57 | v0lZy | phone rings, and the person calling me can hear what i say.... but i cant hear them |
| 13:30.10 | v0lZy | how do i do that from the console? |
| 13:30.19 | [TK]D-Fender | asterisk -rvvvvvvvvvvvvv |
| 13:30.23 | [TK]D-Fender | sip set debug on |
| 13:30.28 | phpboy | definitely a NAT issue _or_ a FW issue |
| 13:31.14 | phpboy | is sip going from the inside out to the provider or from the provider in toward you? connection wise |
| 13:31.55 | v0lZy | well i dont use a password/username |
| 13:32.06 | v0lZy | they just terminate the trunk onto my public IP |
| 13:32.19 | v0lZy | i then NAT 5060 from my public IP to my pbx |
| 13:32.30 | [TK]D-Fender | v0lZy, Show. Us. The. Call. |
| 13:32.34 | v0lZy | and 1:1 NAT a different ip they use for RTP |
| 13:32.41 | [TK]D-Fender | v0lZy, you need a LOT more than just 5060 |
| 13:33.04 | v0lZy | well... i havent tried adding 1:1 on that ip yet... |
| 13:33.16 | phpboy | ah so from the outside in |
| 13:33.17 | [TK]D-Fender | RTP is typically 10000-20000 |
| 13:33.19 | [TK]D-Fender | ^^^^^^6 |
| 13:33.22 | [TK]D-Fender | RTP = audio |
| 13:34.02 | phpboy | v0lZy: is it possible for you to dedicate a public ip to the asterisk server? u can still use nat |
| 13:35.04 | v0lZy | [TK]D-Fender: do i need anything besides 5060 and RTP stuff? |
| 13:35.22 | v0lZy | phpboy: i could put it on a public ip |
| 13:35.31 | v0lZy | i mean... do 1:1 nat |
| 13:35.37 | v0lZy | and they give me a separate IP for it |
| 13:35.51 | [TK]D-Fender | v0lZy, Not for basic SIP |
| 13:35.52 | v0lZy | but i dont see how thats any different than doing 1:1 nat with the ip i alreadyhave |
| 13:35.54 | treborsux | Randomly when a user picks up a call whether it be from outside dahdi or internal extension the user just hears ringing like they made a call. If they hit hold and resume the ringing goes away and they continue the call. |
| 13:36.12 | treborsux | Fender I showed you the call log before and you said it showed normal call |
| 13:36.17 | phpboy | voyeah, basically 1:1 nat would be in your best interest |
| 13:36.18 | v0lZy | [TK]D-Fender: and both need to go both ways right' |
| 13:36.22 | treborsux | all three systems do it now |
| 13:36.24 | phpboy | or at the very least the easy way out |
| 13:36.48 | v0lZy | let me try that |
| 13:36.52 | v0lZy | see if it changes anything |
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| 13:39.23 | v0lZy | ah crap i have to run |
| 13:39.26 | v0lZy | thanks for all the advice |
| 13:39.28 | v0lZy | ill be back tomorrow |
| 13:39.32 | v0lZy | hopefully i can figure something out |
| 13:39.40 | *** join/#asterisk beek_ (~klinebl@pdpc/supporter/bronze/beek) |
| 13:39.48 | phpboy | try 1:1 nat and work your way down from there |
| 13:41.18 | v0lZy | will do |
| 13:41.19 | v0lZy | thanks |
| 13:43.03 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 13:48.15 | snadge | ok i think i finally sent a fax via t38 |
| 13:49.18 | snadge | has someone got a fax number so i can verify that the fax comes through correctly? |
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| 13:57.08 | wdoekes2 | snadge: did my faxnr not work? |
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| 14:13.42 | snadge | wdoekes2: it failed.. maybe i had the wrong number |
| 14:13.54 | snadge | i just sent a 3 page fax to the USA successfully |
| 14:13.57 | snadge | maybe i had the wrong settings |
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| 14:17.52 | wdoekes2 | hm, oh well.. I'll switch the number back to something useful |
| 14:18.06 | snadge | ahh ok |
| 14:18.11 | snadge | well it wasnt answering i think |
| 14:18.17 | snadge | it said timeout.. lemme check my log |
| 14:18.27 | wdoekes2 | wasn't answering? then you definitely had the wrong number |
| 14:18.37 | snadge | ok i'll try one more time just for kicks |
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| 15:00.31 | leifmadsen | Qwell: ping? |
| 15:01.10 | leifmadsen | Qwell: do you just keep a dahdi-linux-kmod-`uname -r`.spec file for every kernel version you're building for? |
| 15:01.11 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85fc.bcn.adamo.es) |
| 15:04.13 | krotos | hei guys, asterisk have the aviability for snmp query=? |
| 15:06.06 | krotos | without external application i mean |
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| 15:12.31 | *** join/#asterisk mintee (~mintee@2001:470:7:a41::2) |
| 15:12.44 | mintee | so... faxdetect isn't detecting. :( |
| 15:12.55 | mintee | using sangoma hardware detection |
| 15:16.43 | snadge | im trying to work out why i can fax australia and usa.. but i get this error when faxing the netherlands |
| 15:17.18 | snadge | sendfax_t38_init: Audio FAX not allowed on channel '%s' and T.38 negotiation failed; aborting. |
| 15:18.29 | *** join/#asterisk SteveWilliams (~chatzilla@220.224.235.78) |
| 15:19.58 | mintee | i have the exact same setup as a previous machine which was receiving faxs no problem |
| 15:20.03 | mintee | this is bizarre |
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| 15:24.13 | mintee | this is fax over a pri too |
| 15:24.15 | mintee | :/ |
| 15:24.17 | mintee | not SIP |
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| 15:29.02 | Micc | finnaly able to upgrade one server to 1.8.9.2 from 1.6. We'll see how it goes. hopefully no blf lock anymore. |
| 15:30.17 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:30.27 | SteveWilliams | Hi All!!! I have got an asterisk pbx and i was able to setup a DID number to ring a SIP extension. What I want is to ring that extension for 1 call only and forward the next call to another SIP extension if the configured extension is busy with a call.... Please help... I am a newbie.... |
| 15:30.30 | leifmadsen | krotos: what do you mean by "snmp query... without external application" ? |
| 15:30.54 | leifmadsen | SteveWilliams: use the DIALSTATUS variable -- also check out the documentation at http://asteriskdocs.org |
| 15:31.28 | leifmadsen | Micc: BLF lock? |
| 15:31.33 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:31.33 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 15:32.35 | SteveWilliams | leifmadsen okay... |
| 15:32.48 | Cadey | hi guys, anyone put a cisco 7965G or 7942G into SIP mode with a non english display like Chinese or Japanies etc |
| 15:32.48 | [TK]D-Fender | SteveWilliams, "core show application chanisavail" <------------- |
| 15:32.59 | [TK]D-Fender | SteveWilliams, what you need instead... |
| 15:33.33 | leifmadsen | [TK]D-Fender: if it's a sip extension he should probably be using DEVICE_STATE() instead -- ChanIsAvail() is not a recommended method |
| 15:33.57 | [TK]D-Fender | leifmadsen, Sure... |
| 15:34.18 | [TK]D-Fender | SteveWilliams, "core show function DEVICE_STATE" |
| 15:34.33 | Micc | Leifmadsen, yeah there was a memory leak in blf state change in 1.6 and there was some locking problems that I think irroot fixed in 1.8.7 or around there. |
| 15:34.36 | SteveWilliams | [TK]D-Fenderokay... checking that in a moment |
| 15:35.01 | leifmadsen | Micc: oh ok good to know -- I'm working on blf stuff with 1.8 as well, but haven't tested heavy enough to run into that type of issue yet |
| 15:35.44 | Micc | leifmadsen, I was hitting almost once a day even with restarts every night. So I'll know in a couple days if it helps. |
| 15:35.57 | leifmadsen | Micc: coolio |
| 15:36.00 | leifmadsen | Micc: keep me posted |
| 15:36.12 | Micc | k |
| 15:37.15 | Micc | we'll be hitting the busiest part of the day here in the next hour or two. I wanted to do this upgrade on the weekend but didn't really have a choice when it started hanging every day. |
| 15:38.06 | Micc | I tried 1.8.5 a while back, but it was a complete failure. This time around I'm not using parking at all. I wrote my own. |
| 15:39.25 | leifmadsen | I know parking went through a major change sometime around 1.8.6 or 1.8.7 |
| 15:40.19 | *** part/#asterisk beek_ (~klinebl@pdpc/supporter/bronze/beek) |
| 15:40.54 | Micc | yeah, it did, which also made it unusable without major dialplan changes for me. I don't use the normal _NXXXXXXXXX,n,Bla I use s,n with a lot of subs and macros. |
| 15:42.26 | Micc | I probably could have fixed that one pretty easy in the parking code, but when I took at look at that code and how its spread all over everywhere, I decided not to even bother. Plus I like mine better, it is more flexible and push the spot you want to put them in. So you save a key. |
| 15:44.23 | Qwell | leifmadsen: no. it all gets built from a 2.x.x.x dahdi-linux tag. I sed it up from a .spec.in |
| 15:44.52 | leifmadsen | Qwell: hmmm |
| 15:44.52 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 15:47.02 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 15:47.42 | Qwell | let's see.. |
| 15:48.33 | Qwell | sed -e "s/@aversion@/${VERSION}/g;s/@arelease@/${RELEASE}_${DISTRO}/g;s/@kversion@/${KVERSION}/g;s/@distname@/${DISTNAME}/g;s/@distver@/${DISTVER}/g" ${RPM_SPECDIR}/${PROJECT}.spec.in > ${RPM_SPECDIR}/${PROJECT}.spe |
| 15:48.41 | Qwell | c |
| 15:49.32 | leifmadsen | Qwell: so I don't quite understand how you use that... looks like it just writes over the same spec file |
| 15:49.39 | Qwell | then I do KVERSION=2.6.18-....el5 ./build.sh |
| 15:49.48 | Qwell | leifmadsen: no, that's sedding from .spec.in into .spec |
| 15:50.00 | leifmadsen | I understand that I think |
| 15:50.03 | Qwell | I have @kversion@ in the .spec.in, in place of where the kernel version goes |
| 15:50.07 | Qwell | lemme show you |
| 15:50.11 | leifmadsen | k |
| 15:50.36 | Qwell | you are using kmodtool, right? |
| 15:50.47 | Qwell | %{!?kversion: %define kversion @kversion@} |
| 15:50.48 | leifmadsen | I think so |
| 15:50.49 | leifmadsen | yes |
| 15:50.58 | leifmadsen | although I don't have @kversion@ in there |
| 15:51.04 | leifmadsen | I think it's just statically defined right now |
| 15:51.07 | Qwell | @kversion@ is just a placeholder |
| 15:51.09 | Qwell | that's why I sed |
| 15:51.10 | leifmadsen | ya |
| 15:57.04 | p3nguin | snadge: I saw you wanting to su asterisk. You aren't supposed to do that, or asterisk user would have a valid shell. Use AGI to run your perl script instead. |
| 15:57.35 | *** join/#asterisk Defraz (~Defraz@69.20.176.132) |
| 16:04.00 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 16:05.25 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 16:18.17 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 16:21.15 | *** join/#asterisk Lantizia (~lantizia@cpc22-stok16-2-0-cust96.1-4.cable.virginmedia.com) |
| 16:22.35 | Lantizia | Lo, like many systems our phone system records all calls to a CDR (in this case it's mysql)... is it possible to dial out in a way that ensures asterisk doesn't note the call? (i.e. I write a star code that uses some option/flag/feature/app present in asterisk so asterisk knows NOT to write a record of the call to the CDR table) ? |
| 16:23.07 | Lantizia | I want a new job and I'm too cheap to use my own phone - but luckily I run our company phone system :D |
| 16:24.29 | *** join/#asterisk abesamthomas (~abesamtho@61.11.125.45) |
| 16:25.49 | p3nguin | lantizia: NoCDR() |
| 16:26.01 | Lantizia | aha! |
| 16:26.33 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 16:27.53 | rrittgarn | afternoon gents |
| 16:29.36 | rrittgarn | whats the best solution for multi context parking? As in each context gets its own parking lot |
| 16:29.58 | [TK]D-Fender | rrittgarn, It's all in the sample config... |
| 16:30.11 | rrittgarn | my apologies, must have missed that |
| 16:30.39 | [TK]D-Fender | rrittgarn, You creat groups and assign accordingly. |
| 16:30.53 | *** join/#asterisk ccesario (~ccesario@187.17.166.162) |
| 16:32.56 | *** join/#asterisk slidesinger (~slidesing@c-174-57-5-70.hsd1.nj.comcast.net) |
| 16:33.09 | Micc | rrittgarn, best solution is to roll your own. but the standard one is pretty good if you use it just right. |
| 16:39.23 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
| 16:46.46 | Micc | seems like my load average is a little higher with 1.8 |
| 16:47.48 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 16:47.55 | Micc | I bet the 1.8.10 update will help because it doesn't do the udptl sockets for every peer. I've tried to say that was a bug for a long time and no one listened. |
| 16:53.56 | *** join/#asterisk slidesinger (~slidesing@c-174-57-5-70.hsd1.nj.comcast.net) |
| 16:55.26 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v007-159.mobile.uci.edu) |
| 17:03.41 | *** join/#asterisk stephendowed (~stephen@197.255.215.252) |
| 17:06.14 | stephendowed | so i try to execute a simple when simple dial and say a number script and on all softphones am getting a declined.any reasons y? |
| 17:07.08 | stephendowed | cos on the asterisk server it shows me that it plays the digits/4.ulaw but am getting declined on the phones |
| 17:09.05 | stephendowed | like anyone here to answer me? |
| 17:09.09 | [TK]D-Fender | " execute a simple when simple dial " <- please completely rephrase this |
| 17:09.19 | [TK]D-Fender | stand show us the call |
| 17:09.20 | [TK]D-Fender | ~pb |
| 17:09.20 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 17:09.21 | [TK]D-Fender | ^^^ |
| 17:09.34 | [TK]D-Fender | stephendowed, and show us the call |
| 17:11.23 | stephendowed | sorry i mean the dial plan just says say digits 4 wen 123 is dialled @fender |
| 17:12.06 | stephendowed | and it actually shows on the server that it is processing correctly..but declined it shown on the sofphone am testing with..both twinkle and 3cx |
| 17:12.13 | WIMPy | stephendowed: Give us the whole story, not some fragments. |
| 17:12.54 | stephendowed | @wimpy that's the whole story |
| 17:13.26 | [TK]D-Fender | stephendowed, Show. Us. The. Call. |
| 17:13.27 | stephendowed | just tested with xlite and it also shows declined... |
| 17:13.28 | [TK]D-Fender | ~pb |
| 17:13.28 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 17:13.29 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 17:17.47 | *** join/#asterisk chasing`Sol (~cS@197.132.216.38) |
| 17:18.33 | stephendowed | am trying to copy it but its running on a vm and i don't know how to do that cos of the mouse integration thingy |
| 17:18.48 | stephendowed | and mind u am a newbie so pardon me gurus lol |
| 17:20.20 | [TK]D-Fender | SSH in from outside the VM |
| 17:21.15 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 17:21.57 | [TK]D-Fender | or if in X use a "copy all" or similar |
| 17:23.27 | p3nguin | Someone threw a big switch that they shouldn't have. The ISP's entire sysloc just went offline. |
| 17:23.33 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 17:23.38 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 17:23.47 | p3nguin | data and voice modems |
| 17:24.52 | [TK]D-Fender | backs away from the big red button .... |
| 17:25.36 | *** join/#asterisk ccesario (~ccesario@187.17.166.162) |
| 17:26.10 | stephendowed | http://pastebin.com/LcvXbE9A |
| 17:26.19 | stephendowed | dats what it looks like guys |
| 17:27.24 | akrohn | phone doesn't seem to be registered |
| 17:27.51 | stephendowed | it is am sure cos i can make calls betwen softphones |
| 17:28.12 | akrohn | but you can't call out or in, yes? |
| 17:28.12 | stephendowed | but to make dem call 123 and hear d digit 4 i get declined on the phones |
| 17:28.27 | WIMPy | stephendowed: Put an Answer() in the beginning, and it may be a good idea to add a Hangup() at the end. |
| 17:28.35 | *** join/#asterisk BarthezZ (~bart@2001:41d0:2:9d0c::2) |
| 17:28.39 | *** join/#asterisk DarthExpeditor (~IceChat9@96-42-133-130.static.trcy.mi.charter.com) |
| 17:30.06 | WIMPy | If you get issues with cut off audio in the beginning, a Playback(silence/1) may be a good replacement for Answer(), or Answer(300) or something. |
| 17:30.14 | [TK]D-Fender | stephendowed, Auto fallthrough, channel 'SIP/1002-0000003a' status is 'UNKNOWN' |
| 17:30.18 | [TK]D-Fender | stephendowed, auto-fallthrough <------------- |
| 17:30.29 | [TK]D-Fender | stephendowed, You ran out of dialplan and it hung up. |
| 17:31.14 | [TK]D-Fender | stephendowed, * probably didn't even have enough time to setup the audio channel before dropping the call like a rock |
| 17:31.24 | stephendowed | thanks wimpy d answer and hangup did the trick |
| 17:31.46 | stephendowed | and what does setting auto fallthrough = yes really do |
| 17:32.46 | WIMPy | Kill your calls when your dialplan ends without a Hangup(). |
| 17:32.53 | [TK]D-Fender | stephendowed, it is standard behavious VS the old 1.2- IVR standard of "running out of priorities on 's'" |
| 17:33.03 | WIMPy | But that can have unexpected effects when some pattern matches. |
| 17:33.24 | [TK]D-Fender | stephendowed, 1.4+ has it where if not set and you run out then they can just dial something more. Which is almost always bad |
| 17:34.02 | stephendowed | i don't currently have it set so what is the default behaviour yes or no |
| 17:34.30 | [TK]D-Fender | stephendowed, Not sure. Go set it to YES and do things properly |
| 17:34.36 | stephendowed | am rily starting to like this asterisk thingy.....especially with this irc channel thumbs up guys |
| 17:34.54 | WIMPy | That will go away with time. |
| 17:34.59 | stephendowed | @fender i set it to yes in the sip.conf file right? |
| 17:35.23 | stephendowed | @wimpy....lmao!!!! so i shouldn't be too happy dis early is what ur sayin? |
| 17:35.26 | [TK]D-Fender | stephendowed, Assumptions + "defaults" = bad. Explicit = good. This work ethic will save you a lot of grief |
| 17:35.52 | [TK]D-Fender | stephendowed, No, this is extensions.conf under [globals] |
| 17:36.08 | WIMPy | I liked it until I really tried to use it. |
| 17:36.43 | stephendowed | and what else can i set in the extensions.conf globals section that can help cos i just started with an empty extensions.conf file |
| 17:37.02 | stephendowed | @wimpy so y u here...to save newbies like us from some grief right? |
| 17:37.34 | WIMPy | Sometimes :-) |
| 17:37.49 | WIMPy | I still learn new things. |
| 17:38.18 | WIMPy | Still lots of things I've never tried. |
| 17:40.23 | stephendowed | well well nice to have u here..but ma question is are there people out there really using this solution, aside all the commercials on their site... |
| 17:40.34 | stephendowed | is it really a successful software? |
| 17:41.12 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:41.15 | WIMPy | I have some installations running. And the users are mostly ok with it. |
| 17:41.27 | WIMPy | I personally am not. |
| 17:42.11 | akrohn | WIMPy, what sort of things bother you? |
| 17:42.18 | akrohn | i'm pretty new at this myself |
| 17:42.38 | [TK]D-Fender | stephendowed, * is big business. Tens of thousands of user. Who knows at this point |
| 17:42.58 | n3hxs | We have an Asterisk IVR runnig handling thousands of calls. I run one personally at home. |
| 17:43.25 | WIMPy | The feel for users. It feels very limited or old scool compated to what we used to have. |
| 17:43.31 | [TK]D-Fender | stephendowed, My company has been using it since 2005, and I started earlier than that |
| 17:43.37 | n3hxs | We have two commercial systems installed based on Asterisk which are solid as a rock. |
| 17:44.17 | stephendowed | @wimpy old skool or complicated.tot all users had to do was make calls and leave complicated to people like u |
| 17:45.03 | WIMPy | Well, making calls is sometimes not that easy :-) |
| 17:45.10 | stephendowed | @n3hxs thanks dude. u just raised up ma faith...i live in africa and people r lookin for a way out of proprietary solutions. |
| 17:45.18 | WIMPy | And sometimes you want to do more than that. |
| 17:45.34 | WIMPy | Like changing between multiple calls or other features. |
| 17:46.24 | stephendowed | i currently do a lot of linux based implementations for people and i want to add asterisk to ma toolkit u know |
| 17:46.32 | stephendowed | so ama be visiting here often guys.... |
| 17:47.00 | stephendowed | @wimpy is dat a limitation with asterisk or just wif whoever handles the configuration |
| 17:47.18 | WIMPy | yes :-) |
| 17:47.49 | WIMPy | For me it's a PITA to have to implement all basic features myself that would otherwise be taken for granted. |
| 17:48.06 | WIMPy | But only being able to use the dialplan can also make it morte complicated for users. |
| 17:48.49 | WIMPy | Think of a three-way call, e.g.. With Asterisk you have to transfer both calls to a conference room and then join yourself. |
| 17:49.20 | WIMPy | And then yu cannot even end it. |
| 17:49.34 | WIMPy | (in a user compatible way) |
| 17:55.28 | stephendowed | @wimpy hmmm there has to be a way around that...am sure |
| 17:55.47 | stephendowed | or is it that we're used to how things normally work and this system just seems different |
| 17:56.49 | WIMPy | Yes, it feels like the 80s are back. |
| 17:58.41 | [TK]D-Fender | Not for most features I've seen & used |
| 17:59.01 | *** join/#asterisk chasing`Sol (~cS@197.132.216.38) |
| 17:59.37 | [TK]D-Fender | " Think of a three-way call, e.g.. With Asterisk you have to transfer both calls to a conference room and then join yourself." <- how is this the way you end up hving to do it? |
| 17:59.43 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 18:00.13 | WIMPy | How else could I do it? |
| 18:00.46 | ChannelZ | My phone does it on its own |
| 18:00.48 | ChannelZ | :P |
| 18:00.54 | WIMPy | Ok, maybe the phone could do it, but the extra delay is likely to cause bad quality/echo. |
| 18:01.02 | [TK]D-Fender | WIMPy, I dunno, I just press the other line key on my phone and say "conf" |
| 18:01.03 | [TK]D-Fender | The end |
| 18:01.18 | ChannelZ | Why? Someone has to mix the audio whether it's a device or Asterisk |
| 18:01.25 | WIMPy | Yes, but that depends on the phone and has nothing to do with Asterisk. |
| 18:01.42 | [TK]D-Fender | WIMPy, In my 7+ 7+ yeafrs of using * I'll say ... "no". |
| 18:02.06 | [TK]D-Fender | WIMPy, I've never seen anything that deserves the title of "phone" that didn't support it |
| 18:02.21 | [TK]D-Fender | WIMPy, Maybe your personal practices are in the 80's ;) |
| 18:02.37 | [TK]D-Fender | "7+ years" |
| 18:02.39 | [TK]D-Fender | Gah... |
| 18:02.54 | WIMPy | No, 90s. That's when you just pressed a button on your phone to tell the switch to do it. |
| 18:03.51 | [TK]D-Fender | \o/ |
| 18:03.54 | WIMPy | And the whole call forwarding thing is also rather unfortunate. |
| 18:04.02 | [TK]D-Fender | Which? |
| 18:04.17 | [TK]D-Fender | 1 botton on phone. Enter number. Another button on phone. The End |
| 18:04.39 | WIMPy | Either you have to use ugly feature codes or have to use deflection by the phone, but that's not the same. |
| 18:05.40 | *** join/#asterisk pdtpatr1ck (~pdtpatric@12.249.4.226) |
| 18:05.51 | WIMPy | If you have an extension that rings two phones, uing the phones feature is a no go. |
| 18:06.31 | [TK]D-Fender | WIMPy, IIRC there is a Dial option to reject redirects like that. |
| 18:06.49 | WIMPy | There is. |
| 18:07.03 | WIMPy | So we're back to having to use the dialplan. |
| 18:07.42 | WIMPy | And if you want a common status display, you have to (ab)use BLF functionality for that. |
| 18:07.42 | *** join/#asterisk abesamthomas (~abesamtho@61.11.125.45) |
| 18:07.56 | [TK]D-Fender | if you want to avoid the redirect killing it... there is a 2nd option : dialing via nested local channel |
| 18:08.01 | WIMPy | And that won't even tell you the type or destination of a forward. |
| 18:08.32 | WIMPy | Hmm. How does that help? |
| 18:08.52 | [TK]D-Fender | WIMPy, Yes making the status visible on the phone is trickier. Different phone offer more choices. Polycom's MB is way, mseveral others have their own. |
| 18:08.57 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 18:09.24 | [TK]D-Fender | WIMPy, by dialing nested local channels each device is it's own Dial, so the redirect on one doesn't kill the group. |
| 18:09.58 | WIMPy | Got you. Yes, but that is another function. |
| 18:10.20 | [TK]D-Fender | What is another function? |
| 18:10.22 | WIMPy | Redirecting one phone of an extension vs redirectiong the extension. |
| 18:10.50 | [TK]D-Fender | Well yes separating the concept of a physical phone VS one means of contacting it. |
| 18:11.06 | [TK]D-Fender | I've never seen any system that let you do that as a separate entity anyway... |
| 18:11.22 | WIMPy | I'm used to exactely that. |
| 18:11.50 | [TK]D-Fender | WIMPy, You are weird and need to be fixed ;) |
| 18:12.10 | WIMPy | It's important. Think about immediately redirecting the whole extension to VM when the last one leaves the office. |
| 18:12.22 | WIMPy | No, I'm just not stuck in the 80s. |
| 18:12.47 | [TK]D-Fender | What is "whole extension" vs "everyone leaving the office"? |
| 18:12.49 | WIMPy | But as I learned here that's a geographocal thing. |
| 18:13.09 | [TK]D-Fender | Sounds like you should be using a queue for this sort of thing anyway |
| 18:13.16 | Micc | WIMPy, I use a special night mode blf for that kind of thing. they press it when the last one leaves. |
| 18:13.22 | WIMPy | Just an exaple to make the importance of the difference obvious. |
| 18:13.52 | [TK]D-Fender | WIMPy, It may be an example... it doesn't mean it isn't a retarded and shitty one ;) |
| 18:13.58 | WIMPy | You couldn't do that on the phone, even if you used local channels. |
| 18:14.46 | Micc | WIMPy, I'm not following what your trying to do. Can you explain it again? I'm pretty sure I can do just about anything with asterisk and enough time. |
| 18:14.56 | WIMPy | Well, you could, but not with the desired result. |
| 18:15.22 | WIMPy | Sure you can, but it's neither easy for the admin nor for the user. |
| 18:17.13 | WIMPy | The forwarding thin even works better without any PBX at all. |
| 18:17.17 | [TK]D-Fender | WIMPy, little dialplan feature is "easy" for the user. Anyone who really resists on that one is pretty much useless. There is a complication for visibility of now-server-based functionality you implement this way. BLF is one tool, and browser capable phones are another to add more. |
| 18:18.15 | [TK]D-Fender | WIMPy, Considering Polycom IP32X/33X offer this starting at ~$80 USD ... it's typically a moot point for the user and end budget. Just a little more trickery for the admin. But then again.. what yare you paying them for? :) |
| 18:19.06 | WIMPy | Browsers can do it, yes, but they are different for every phone. |
| 18:19.35 | [TK]D-Fender | yes, which is why you should just settle on Polycom like so many of us happily do :) |
| 18:20.15 | WIMPy | I've never come across one. They don;t seem very popular over here. |
| 18:20.17 | [TK]D-Fender | WIMPy, Are you dealing with tons of users who are picking theirs all over the place and putting the support burden on you? |
| 18:21.30 | WIMPy | Luckily no, but a good system should work with any brand of phone. |
| 18:22.41 | Kobaz | anyone have an example of polycom corp directory via ldap |
| 18:22.58 | Micc | WIMPy, have you looked at the yealink T38G? |
| 18:25.18 | [TK]D-Fender | WIMPy, Well "good system" and "work they way you specifically want" are not compaitble concepts. * is agnositc. Every phone supports a different featureset and levels of compatability. For what you want you shuold jsut settle on a proprietary solution and pay the price |
| 18:27.13 | *** join/#asterisk blizzow (~jburns@74.7.49.235) |
| 18:28.06 | WIMPy | Micc: Haven;t seen one IRL, but the do look interesting. |
| 18:29.07 | Micc | WIMPy, we're using them for all our new customers. We used to do a lot of aastra and polycom, but for the price point you get a good phone with lots of features, color screen and gigabit. |
| 18:29.32 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 18:29.32 | WIMPy | [TK]D-Fender: I'd certainly go for the classic PBX. That's probaly not even more expensive than the Asterisk way. |
| 18:29.37 | Micc | lots of programmable buttons too. |
| 18:30.03 | WIMPy | In fact I have charegen mor than a propritary solution would cost on top of that. |
| 18:30.35 | [TK]D-Fender | WIMPy, Now if you weren't so picky you could be a much more fiscally sound and happy person ;) |
| 18:30.42 | WIMPy | Damn. I really need to learn to use the keybaord again, I think :-( |
| 18:31.00 | [TK]D-Fender | WIMPy, Don't worry, my typing is slowing going to shit as well... |
| 18:31.27 | WIMPy | Is that an Asterisk induced desease? |
| 18:31.49 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 18:32.25 | WIMPy | Well, as I said: I earned more by using Asterisk, but the result is not what I'd prefer it to be. |
| 18:34.32 | Micc | WIMPy, once you impliment those features in the dialplan once don't you reuse that for all your customers? |
| 18:34.58 | Micc | I guess it would be harder if its not a multi-tenant system. |
| 18:34.59 | [TK]D-Fender | Micc, MY WHEEL WILL BE ROUNDER DAMMIT! AND WITH SPRINKLES THIS TIME! |
| 18:35.08 | WIMPy | If they wouldn't have special needs, they probably wouldn;t pay me at all :-) |
| 18:35.09 | Micc | lol |
| 18:35.12 | [TK]D-Fender | <PROTECTED> |
| 18:35.59 | Micc | I'd hate to have to impliment special features for over 150 customers if they all had their own asterisk system on premise. |
| 18:36.44 | Micc | but I still do a lot of custom stuff. Whatever they can think up can usually be done somehow. |
| 18:37.37 | WIMPy | The good thing about customers is that they are usually happy with whatever you give them. |
| 18:38.18 | Micc | thats usually true, and more true when they have a really bad system to start with. |
| 18:39.18 | WIMPy | That doesn't happen to me. |
| 18:39.42 | Micc | We always have to do a little training to make sure they know how to do things. But once they get it they usually like it better than the way they were doing things. |
| 18:39.54 | WIMPy | But the classic systems are bad at things like announcements. |
| 18:42.02 | [TK]D-Fender | WIMPy, You seem to have become our karmic effigy. All the bad stuff happens to you so the rest of us can enjoy *'s good fortunes :) We thank you for your unwitting role.... |
| 18:43.58 | WIMPy | If I still had a real phone line, I'd probably dedust my 15 year old plasic PBX and use that again. |
| 18:44.30 | WIMPy | But I don't so the available features are very limited anyway. |
| 18:52.03 | Micc | So, let me get this right, you can do everything you want with asterisk, but you don't want to have to write dialplan code to make it work? |
| 18:54.00 | *** part/#asterisk jeffkap (~jkaplan@vega.jeffkaplan.net) |
| 18:55.10 | [TK]D-Fender | "custom shortcuts" d-bus |
| 18:55.17 | [TK]D-Fender | oops, wrong window.. |
| 18:58.01 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 19:01.57 | karlfife | When I reboot my asterisk server, I can not set up calls on the local channel until I reload the dialplan once. Then it runs perfectly forever until I reboot again. This started somewhere mid-1.6.2. I never tried to fix it because 1.6.2 is EOL. Yesterday I migrated to 1.8.9.2 and the problem is still there. Anyone heard of this? |
| 19:03.15 | karlfife | Also, don't accidentally uncomment the first line of res_fax_digium.conf or asterisk will segfault :-) |
| 19:03.28 | p3nguin | Put into modules.conf: preload => chan_local.so |
| 19:03.41 | p3nguin | Then restart asterisk to see if it is cured. |
| 19:03.46 | karlfife | Brilliant. |
| 19:03.49 | karlfife | Let me try this... |
| 19:06.24 | karlfife | Waiting for a call to hang up :-) |
| 19:06.49 | karlfife | p3nguin: any thoughts as to why this would have changed mid 1.6.2 branch? |
| 19:07.14 | karlfife | I don't believe I ever had this issue befor then |
| 19:07.43 | karlfife | CLI> reboot when convenient :-) |
| 19:08.05 | p3nguin | Not really. I always suffered from a similar problem when using local channels from app_queue, so I've preloaded chan_local and pbx_config for a very long time. And I never used any of the 1.6.x brances (I only use LTS branches). |
| 19:20.30 | karlfife | p3nguin: Thanks. We only went to development branches because there were some essential features for us. Now that they're in LTS we're stoked. The next-gen essential features we will turn up on 1.10 VM's using the LTS instances as media gateways. |
| 19:22.37 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 19:33.07 | karlfife | p3nguin: Hmmmm. Didn't seem to help |
| 19:33.15 | karlfife | chan_local.c:899 local_call: No such extension/context @outbound-mobile while calling Local channel |
| 19:33.15 | karlfife | -- Couldn't call Local/@outbound-mobile/n |
| 19:33.30 | karlfife | but after a dialplan reload... it can. |
| 19:35.09 | p3nguin | Are you omitting the extension for pasting purposes, or is that verbatim? |
| 19:35.32 | karlfife | Good point. |
| 19:35.35 | karlfife | Verbatim. |
| 19:35.40 | p3nguin | That's a problem. |
| 19:35.41 | karlfife | Let me see what it tries to call when I reload |
| 19:36.01 | [TK]D-Fender | karlfife, -- Couldn't call Local/@outbound-mobile/n <-------- I don't see the EXTENSION in there... |
| 19:36.05 | p3nguin | Whatever is creating that local channel doesn't know there's supposed to be an extension. |
| 19:36.14 | karlfife | right. |
| 19:36.25 | karlfife | It is as if a global variable is not being loaded until reload |
| 19:36.42 | karlfife | [TK]D-Fender: bingo! |
| 19:36.43 | p3nguin | It should be loaded as soon as pbx_config comes up. |
| 19:37.17 | [TK]D-Fender | karlfife, maybe you should be a little more thorough in showing us what you're doing. |
| 19:38.06 | karlfife | Dial("SIP/140.239.58.227-0000000d", "SIP/201&SIP/203&SIP/205&Local/3125656566@outbound-mobile/n,30,r") |
| 19:38.51 | [TK]D-Fender | karlfife, How about the COMPLETE call attempt with the error... and the dialplan ... and variable dumps, etc |
| 19:38.58 | [TK]D-Fender | ~pb |
| 19:38.58 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 19:38.59 | [TK]D-Fender | ^^^^^^^^^^^^^6 |
| 19:39.28 | karlfife | [TK]D-Fender: ^^^^^^^^^ 3 lines. I know. |
| 19:39.33 | karlfife | Sheesh. |
| 19:40.16 | p3nguin | I hate using Dial's r option, but I've recently had to add it in a lot of places. Several calls have been silent while dialing, and that's not acceptable to people who don't understand the underlying technology. |
| 19:40.38 | karlfife | p3nguin: me too. |
| 19:40.43 | p3nguin | Luckily, the ones I have monitored have just waiting through the silence. |
| 19:40.51 | p3nguin | s/waiting/waited/ |
| 19:41.20 | p3nguin | 30 seconds or silence is a long time when you think a call is supposed to be progressing. |
| 19:41.28 | p3nguin | s/or/of/ |
| 19:41.44 | karlfife | Amen my brother. Also, it doesn't seem to ever happen unless bridging TDM to sip. |
| 19:42.36 | karlfife | So the obvious problem in my case is that is that somehow the variable is not being set initially. It's a simple "foo=1234" in a static dialplan |
| 19:42.46 | p3nguin | I've only noticed it when calling an SCCP phone, but it happens if the call comes into the system via SIP or IAX2. |
| 19:43.15 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 19:43.32 | *** join/#asterisk dxd828 (~dxd828@88-109-112-31.dynamic.dsl.as9105.com) |
| 19:43.36 | *** part/#asterisk dxd828 (~dxd828@88-109-112-31.dynamic.dsl.as9105.com) |
| 19:44.10 | karlfife | I guess them's the brakes when you're marrying differnt technologies. God bless it/god d@m# it |
| 19:47.11 | karlfife | p3nguin. |
| 19:47.19 | karlfife | Good eye. |
| 19:47.33 | karlfife | It seemed so obvious to me as not to even look there. |
| 19:47.49 | karlfife | Where's the remote? It's in my hand. Ooops. |
| 19:47.56 | karlfife | So here's what I think: |
| 19:48.12 | karlfife | static variables |
| 19:48.12 | karlfife | KCELL =3125656566 |
| 19:48.36 | karlfife | xKCELL =Local/$[${KCELL}]@outbound-mobile/n |
| 19:48.50 | karlfife | the dialplan is dialing xKCELL |
| 19:48.59 | mentax | Hi all |
| 19:49.22 | p3nguin | Very inter-nesting. |
| 19:49.32 | [TK]D-Fender | karlfife, WTF is your variable reference doing wrapped in a EXPRESSION? |
| 19:49.51 | karlfife | not allowed? |
| 19:49.53 | [TK]D-Fender | p3nguin, Very punny... |
| 19:49.57 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 19:50.05 | [TK]D-Fender | karlfife, it isn't an expression. You just want the value |
| 19:50.38 | karlfife | What's the proper way to next a value like that? |
| 19:51.01 | mentax | Can someone help me with my problem? When somebody call to my asterisk configuration, he doesn't hear call tone, but my phone ringing and I can answer it. If I no answer - it send people to voice mail |
| 19:51.12 | [TK]D-Fender | ${VAR} |
| 19:51.13 | mentax | But I don't hear tone =( |
| 19:51.43 | [TK]D-Fender | mentax, make sure you have a proper indications.conf in place. |
| 19:52.27 | mentax | [TK]D-Fender: I use freepbx, I check it, and I have it in /etc/asterisk |
| 19:52.51 | [TK]D-Fender | mentax, then pastebin the complete call with SIP DEBUG (if applicable) enabled. |
| 19:52.52 | [TK]D-Fender | ~pb |
| 19:52.52 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 19:52.54 | [TK]D-Fender | ^^^ |
| 19:55.01 | mentax | [TK]D-Fender: http://pastebin.com/i31m5Ych |
| 19:55.31 | mentax | [TK]D-Fender: I use sip trunk |
| 19:56.55 | *** join/#asterisk abesamthomas (~abesamtho@61.11.125.45) |
| 19:57.19 | karlfife | [TK]D-Fender: I see. Wrong too. Do I need t use the EVAL funciton since otherwise the parser otherwise won't evaluate the nested variable? |
| 19:57.34 | karlfife | too/tool |
| 19:59.17 | Qwell | karlfife: ${${FOO}} works fine |
| 19:59.26 | Qwell | assuming ${FOO} contains BAR, and ${BAR} exists |
| 20:00.42 | [TK]D-Fender | karlfife, it isn't nested. You refernced your variable inside and expression. There is no need to. |
| 20:00.53 | [TK]D-Fender | Oh wait... |
| 20:00.54 | karlfife | where FOO=you, ${Thank ${FOO}} |
| 20:02.16 | karlfife | Seriously. Thanks. |
| 20:02.22 | [TK]D-Fender | karYou may need EVAL() for that... and you still don't need an expression |
| 20:02.23 | Qwell | karlfife: Just send beer. |
| 20:02.51 | *** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de) |
| 20:03.21 | karlfife | I tried a bunch of things years ago when I wrote it. I remember being surprised that $[] worked. Still surprised taht it works after reloading the dialplan. |
| 20:03.44 | p3nguin | /var/lib/asterisk/sounds/en/extra-sounds-en.txt indicates that there should be a file named "callerid" but it does not exist. Anyone know why? |
| 20:03.55 | mentax | [TK]D-Fender: any idea? |
| 20:04.14 | [TK]D-Fender | mentax, Idea : provide what I asked for |
| 20:04.35 | karlfife | Ask Allison! |
| 20:05.06 | p3nguin | I see ha/callerid. Maybe that's what it meant. |
| 20:05.28 | karlfife | She'll know why that file's missing! Or just place your own recording, preferably a deep male voice. |
| 20:05.39 | file | is missing? |
| 20:06.37 | [TK]D-Fender | In Soviet Russia file 404's YOU |
| 20:14.23 | mentax | [TK]D-Fender: http://pastebin.com/it27Zhy4 |
| 20:15.54 | [TK]D-Fender | mentax, <--- Transmitting (no NAT) to 208.64.8.13:5060 ---> SIP/2.0 180 Ringing |
| 20:16.13 | [TK]D-Fender | mentax, OK< your system is sending them ringing status but they don't seem to pass it to the person originating the call. |
| 20:16.31 | [TK]D-Fender | mentax, Click on the "answer this call" option for your inbound route |
| 20:16.52 | [TK]D-Fender | mentax, That should force the audio to be inband from the "r" dial option you are using there |
| 20:18.07 | *** join/#asterisk exothermc (~exothermc@m6.office2-ww.wideideas.net) |
| 20:19.05 | exothermc | Has anyone done any simple integration with asterisk and salesforce.com? I'm not looking for a desktop application, just a asterisk based solution that logs calls into the correct lead/account if they exist. |
| 20:20.00 | leifmadsen | exothermc: if there is an API interface for salesforce then it'd be as simple as using it via an AGI |
| 20:20.10 | [TK]D-Fender | Or DB lookup |
| 20:20.13 | leifmadsen | then you'd just pass the required data to the salesforce api |
| 20:20.24 | leifmadsen | yes, or DB if you have direct access to the DB, but an API is safer |
| 20:20.48 | leifmadsen | "logs calls into the correct lead/account" implies writing |
| 20:20.52 | [TK]D-Fender | DB will make you look cooler for not needing it gift-wrapped and going in and taking what you want :) |
| 20:21.11 | leifmadsen | writing to the DB directly is more likely to cause corrupted data |
| 20:21.21 | leifmadsen | I don't see that being the "cooler" way |
| 20:21.58 | exothermc | Always a fun to ask "Has anyone invented the wheel" and have many people state that inventing the wheel is possible. |
| 20:22.32 | leifmadsen | exothermc: as I've never done it, I'm telling you how I would approach it. If you're looking for the "has anyone done X" google is likely a better source of information |
| 20:22.32 | [TK]D-Fender | exothermc, the trick is to rework the definition to devalidate prior art ;) |
| 20:23.43 | *** join/#asterisk mistermocha (~Adium@173-164-169-21-SFBA.hfc.comcastbusiness.net) |
| 20:24.04 | mistermocha | hey all… more of a hardware question |
| 20:24.10 | mentax | TK]D-Fender: how can I do this? |
| 20:24.33 | mistermocha | I have a polycom soundstation2 here from our old phone system |
| 20:24.52 | mistermocha | we're moving to voip now… I thought it was an IP 6000 |
| 20:25.04 | exothermc | [TK]D-Fender: good call. |
| 20:25.21 | mistermocha | now I'm trying to find out if I can make it work over VoIP, or if I need to replace the phone |
| 20:25.42 | mistermocha | a new IP 6000 is upwards of $600 |
| 20:26.34 | leifmadsen | mistermocha: you'll need an ATA |
| 20:26.39 | leifmadsen | soundstation2 is analog only |
| 20:26.40 | _Corey_ | mistermocha: You can get an ATA like a PAP2 for $50 |
| 20:28.29 | WIMPy | Micc: No I want to be able to do it in a comfortable and uniform way as a user. |
| 20:29.03 | WIMPy | But as a admin I don't like having to invent the wheel myself, eiter. |
| 20:29.15 | WIMPy | I only do it for the money. |
| 20:41.33 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 20:48.59 | *** join/#asterisk TheCompWiz (~TheCompWi@wsip-68-109-200-102.mc.at.cox.net) |
| 20:49.16 | TheCompWiz | anyone know what would cause this: "-- Connected line update to SIP/1098001-00000014 prevented." ... when adding video to a call... |
| 20:59.13 | leifmadsen | could have something to do with ccss |
| 20:59.20 | leifmadsen | guesses wildly |
| 21:10.31 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:18.21 | *** join/#asterisk Scar-G (enforcer@182.178.29.196) |
| 21:18.47 | Scar-G | Hello everyone |
| 21:19.56 | Scar-G | What should I replace 'x' for executing my command on all channels ( AMI question ) |
| 21:19.57 | Scar-G | Action: PlayDTMF |
| 21:19.58 | Scar-G | Channel: x |
| 21:19.58 | Scar-G | Digit: 1 |
| 21:20.36 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 21:21.45 | [TK]D-Fender | Scar-G, there is no "all channels" |
| 21:21.52 | [TK]D-Fender | Scar-G, 1 by 1 |
| 21:23.34 | Scar-G | Oh goodie. |
| 21:23.53 | Scar-G | I will have to perform 1 by 1 for 100+ channels then ? |
| 21:23.57 | [TK]D-Fender | yes |
| 21:24.52 | Scar-G | perhaps some script will help here ? |
| 21:24.59 | [TK]D-Fender | Probably |
| 21:26.07 | Scar-G | Can you explain how to ? |
| 21:27.10 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 21:27.22 | [TK]D-Fender | Scar-G, Go get the channel list. Pick the ones you want. do what you want with them |
| 21:28.03 | Scar-G | How to go to the channel list ? |
| 21:28.10 | Scar-G | using GUI ? |
| 21:28.21 | p3nguin | Asterisk has no GUI. |
| 21:28.36 | [TK]D-Fender | AMI <--------- |
| 21:28.52 | Scar-G | Action: Channel List ? |
| 21:29.33 | [TK]D-Fender | What does it say it does? |
| 21:29.56 | [TK]D-Fender | checkout time, BBIAB |
| 21:36.18 | *** join/#asterisk magicrhesus (~magicrhes@aether.hipocoon.be) |
| 21:37.50 | *** join/#asterisk revolve (~3@cpc1-stre1-0-0-cust1001.1-1.cable.virginmedia.com) |
| 21:37.54 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 21:51.52 | Micc | so far so good with 1.8.9.2. Had a bit of a high load average for a while, but everything seemed to be running smoothly. |
| 21:52.31 | Micc | Lots of phones doing a lot of blfs, no dead lock yet. |
| 22:01.03 | *** join/#asterisk funkylonehat (~funkylone@125-236-222-73.adsl.xtra.co.nz) |
| 22:05.58 | Micc | It might even be good enough for me to go snowboarding tomorrow. |
| 22:06.32 | akrohn | i just upgraded out 1.6.0.6 box to latest 1.6 a couple weeks ago. no crashes or dead locks since! |
| 22:06.35 | akrohn | our* |
| 22:06.47 | Micc | akrohn, do you do a lot of blfs? |
| 22:06.57 | akrohn | haha no idk what they are |
| 22:07.09 | Micc | then you should be fine with 1.6 |
| 22:07.32 | akrohn | oh linux from scratch? |
| 22:08.04 | akrohn | i would like to eventually. once i get the current system nailed down, i'd like to build a custom linux kernel and os to run * on |
| 22:08.31 | akrohn | but i might just be reinventing a wheel or three |
| 22:10.57 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 22:12.21 | malcolmd | heh |
| 22:14.11 | malcolmd | busy lamp fields, for those playing along at home |
| 22:15.43 | *** join/#asterisk Nasga (~Nasga@AAmiens-157-1-100-145.w86-208.abo.wanadoo.fr) |
| 22:16.13 | Micc | akrohn, you can run asterisk on a bunch of different kernels and devices. You can even run it on a router. |
| 22:16.47 | Micc | akrohn, there are already some distributions that have linux + asterisk already in one package. |
| 22:19.05 | leifmadsen | Micc: oh that's good news about the blf's |
| 22:20.47 | Micc | leifmadsen, yeah so far so good. I might not have to baby sit the servers as much now. :) |
| 22:21.11 | leifmadsen | Micc: excellent. I'm going to be using BLF heavily soon, so it's good for someone else to have tested for me ;) |
| 22:22.33 | Micc | I can't say its better yet, but its certainly no worse than it was. |
| 22:22.37 | Micc | another day or two will tell. |
| 22:22.52 | ChrisInSydney | Just jumped in, whats the BLF issue ?? |
| 22:23.01 | Micc | I'm watching the CLI and I see state changes like 20 or 30 every few seconds. |
| 22:23.25 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:23.42 | Micc | ChrisInSydney, I always got deadlocks with 1.6. I'm testing 1.8 now to see if it fixes my deadlock with blf problem. |
| 22:24.12 | ChrisInSydney | Micc: What phones ? |
| 22:24.29 | ChrisInSydney | I had issues with the T1 timing on Snom370s with Sidecars |
| 22:24.34 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 22:24.52 | Micc | I've got all different kinds of phones. Mostly aastra, polycom and yealink. |
| 22:25.03 | Micc | 272 devices. |
| 22:25.36 | ChrisInSydney | OK, all with HINT "extensions" in the dialplan |
| 22:25.45 | Micc | yes. |
| 22:26.06 | ChrisInSydney | I had some strange issues with the same in 1.4.22. similar |
| 22:26.20 | ChrisInSydney | unloaded chan_sip and reloaded and it all came good |
| 22:26.21 | ChrisInSydney | wierd |
| 22:26.38 | ChrisInSydney | if you did a relaod, it would just take forever |
| 22:26.42 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:27.16 | ChrisInSydney | somewhere over 200 extensions, cant remember exactly |
| 22:28.42 | Micc | ChrisInSydney, you had it deadlock and fixed it with a module reload chan_sip? Or module unload chan_sip, then module load chan_sip? |
| 22:28.56 | ChrisInSydney | module unload |
| 22:29.03 | ChrisInSydney | module load |
| 22:29.11 | ChrisInSydney | yup |
| 22:29.18 | Micc | thats an interesting idea. I should try that next time. |
| 22:29.26 | Micc | I figured it was deadlocked in the core. |
| 22:29.52 | ChrisInSydney | the service was on a VPN and someone something wasnt playing fair |
| 22:29.54 | Micc | Can you actually unload when there are a bunch of channels open? |
| 22:30.00 | ChrisInSydney | but no AMI interface |
| 22:30.17 | Micc | When it dead locks for me I've got 20-30 calls going on. |
| 22:30.27 | ChrisInSydney | had to, they didnt like it, but they didint like what was going either |
| 22:31.01 | ChrisInSydney | call centre ?? |
| 22:32.02 | Micc | no, just a bunch of customers. |
| 22:32.11 | Micc | multi-tenant system. |
| 22:32.13 | ChrisInSydney | same here |
| 22:33.32 | Micc | I could never upgrade before now because 1.8 does parking differently and it didn't work right for multi-tenant. |
| 22:33.47 | Micc | So I rolled my own, and its even better than the original. |
| 22:33.59 | Micc | only about 30 lines of dialplan and some database. |
| 22:34.03 | ChrisInSydney | same here, got it working on 1.4 a few years ago |
| 22:34.17 | ChrisInSydney | using extensionstate ? |
| 22:34.28 | Micc | you mean devicestate? |
| 22:34.48 | ChrisInSydney | one of those :-/ |
| 22:34.58 | ChrisInSydney | its been a wile |
| 22:35.00 | ChrisInSydney | while |
| 22:35.38 | Micc | yeah, custom blf hints with devicestate |
| 22:35.58 | Micc | and its one touch park, just hit the spot you want it to go into. |
| 22:36.18 | Micc | so you don't need a separate park button. |
| 22:36.37 | Micc | I would share, but I don't think my partner would like that. |
| 22:39.17 | rrittgarn | eplains why you were telling me to just write my own earlier Micc |
| 22:39.39 | rrittgarn | explains even. I'm still having multi-tennant issues with my park setup as is... |
| 22:40.58 | Micc | rrittgarn, because I don't like the parking built into asterisk. The code for it is all over the place to make it work properly. Its much cleaner to just do it with a few lines of dialplan and func_odbc. |
| 22:41.11 | ChrisInSydney | Micc: Thats cool. Have bee able to get that working on Cisco SPA and Aastra. haven't played with Yeakink |
| 22:41.25 | ChrisInSydney | thats cool |
| 22:41.46 | *** join/#asterisk jsjc (~Adium@161.Red-83-45-143.dynamicIP.rima-tde.net) |
| 22:42.17 | ChrisInSydney | Micc: So if you have a call, you press the park 1 button, the call goes away and park 1 lights up |
| 22:42.23 | ChrisInSydney | press park 1 again and retrieve the call |
| 22:42.29 | Micc | yes, correct. |
| 22:42.29 | ChrisInSydney | ?? |
| 22:42.32 | ChrisInSydney | cool |
| 22:42.40 | ChrisInSydney | on Yealink, Aastra and Polycom |
| 22:42.46 | Micc | yes |
| 22:43.06 | Micc | and it would work on virtually any phone that works with asterisk and has blf transfer. |
| 22:43.47 | rrittgarn | so for the BLFs as you were saying you just have your dialplan set the device state on those specific hints, thus lighting everything up... Do you just keep track of who is where with Func_ODBC? |
| 22:43.48 | ChrisInSydney | So the phone needs to be able to do a BLF transfer. ie Press BLF and call gets blind transferred to BLF extension |
| 22:45.41 | ChrisInSydney | You using custom hints, or hints from the parking lot ? |
| 22:48.26 | Micc | ChrisInSydney, correct. custom hints. |
| 22:48.34 | Micc | rrittgarn, yes. |
| 22:49.13 | ChrisInSydney | ahh, maybe where things are commng unstuck :-/ ?? |
| 22:49.55 | Micc | rrittgarn, it gets a little complicated with shared variables if you're pre 1.8 because of a problem with bridge or how it executes dialplan after a channel is dead. |
| 22:50.34 | ChrisInSydney | I stuck back in 1.4 land as 1.6 used to do wierd things on me. Then I just forgot about anything else. Finally figured 1.8 needs a good look as 1.10 is out ;-) |
| 22:54.19 | rrittgarn | nah I'm on 1.8.8.1 |
| 22:54.40 | rrittgarn | So among my many other pieces to this puzzle I'll have to add this I guess... you have any references you can share? haha |
| 22:55.29 | Micc | sorry, I wish I could, but my biz partner would kill me. |
| 22:56.36 | Micc | It took me about 8 hours to figure out once I got the idea. |
| 22:56.52 | Micc | at least you know it can be done. |
| 23:06.16 | autofsckk | hi, im testing with the echo test but i dont have any audio, i use normally ulaw but at the CLI i see is using gsm, is theere a way to mke it use alaw? it was working right, but my provider told me to put disallow= all and allow=all al [general] on sip.conf, i already deleted that, but it still puts gsm first |
| 23:07.37 | ChrisInSydney | <autofsckk>: Disalo=all; allow=ualw; allow=alaw; allow=gsm will push the GSM codec down the preferred list |
| 23:08.00 | ChrisInSydney | you also need to have the files loaded, in the right DIR. Had this issue the otehr week |
| 23:08.08 | ChrisInSydney | sound files |
| 23:23.23 | *** join/#asterisk Dlukz (~dlukz@dlukz.dlukz.com) |
| 23:24.13 | Dlukz | I'm trying to run ExecIf on Asterisk 1.4.27.1-1 and I cannot seem to get it to work |
| 23:24.54 | Dlukz | anyone experianced with execif? |
| 23:26.45 | *** join/#asterisk xpot-mobile (~james@dhcp68.emcb.utah.edu) |
| 23:33.11 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 23:36.51 | autofsckk | ChrisInSydney: i installed the ulaw sound files, i thought i wasnt getting any audio because of the gsm codec, but what else could be causing this? it stopped working, im trying to make it work with my provider account, and he told me to put disallow=all allow=all and after that it just stopped working, well i dont hear anything anymore |
| 23:38.43 | ChrisInSydney | in 1.8 you need to have the sound files in the /var/lib/asterisk/sounds/en |
| 23:39.09 | ChrisInSydney | do disalow=all; allow=ulaw; and see what happens |
| 23:39.15 | autofsckk | i have them installed |
| 23:39.40 | autofsckk | it now loads it the right way |
| 23:39.52 | autofsckk | -- <SIP/0014BFFD36F5-pap-00000001> Playing 'demo-echotest.alaw' (language 'es') |
| 23:41.46 | autofsckk | well it sounds now with twinkle, but not with the pap :S |
| 23:45.15 | ChrisInSydney | autofsckk: The handset will make a difference |
| 23:46.00 | ChrisInSydney | try dissalow=all; allow=g722; ;-) |
| 23:48.44 | rrittgarn | anybody have any experience on getting Avaya 5410's working with Asterisk? Just had a customer ask... |
| 23:50.09 | p3nguin | Anyone who instructs to put both disallow=all and allow=all in the same section is an idiot. |
| 23:50.24 | *** join/#asterisk rajiv_ (~rajiv@gentoo/developer/rajiv) |
| 23:50.43 | p3nguin | dlukz: What's your issue? I can help you with it. |
| 23:55.18 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 23:55.50 | *** join/#asterisk talntid (~t@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 23:56.38 | ChrisInSydney | <p3nguin> Anyone who instructs to put both disallow=all and allow=all in the same section is misinformed and probably too soon |
| 23:56.53 | ChrisInSydney | hey p3nguin, hows life ? |
| 23:57.17 | p3nguin | I... didn't say that sentence... |
| 23:58.14 | ChrisInSydney | should have sead spoke too soon |
| 23:58.45 | ChrisInSydney | I am just being lenient |
| 23:59.05 | ChrisInSydney | But we aprobably both right i some wasys. A misinformed idiot |
| 23:59.07 | p3nguin | I'm confused. You're quoting me, but I didn't say that text. |
| 23:59.56 | ChrisInSydney | thats OK. I quoted you quoting the misinformed idiot |