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| 00:12.25 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 00:29.34 | snivets | Does anyone know why Mitels display the "Page Not Found" line on the top of the screen, and what can be done about it? |
| 00:35.56 | rrittgarn | having a bit of an issue keeping a channel alive. I'm using a macro on a dial out [Dial(SIP/outbound,30,M(Press1toAccept))], to ask the user if they want to take the call or not, if they hang up, the whole call drops instead of just their channel. Any easy solution here? |
| 00:37.58 | [TK]D-Fender | rrittgarn: "core show application dial" <- |
| 00:39.14 | rrittgarn | Whats funny is this is a macro you wrote Fender |
| 00:40.19 | rrittgarn | and ty for pointing me to the obvious... F is probably the option im looking for. |
| 00:40.59 | rrittgarn | or maybe not... |
| 00:46.35 | [TK]D-Fender | g <- |
| 00:47.18 | [TK]D-Fender | heads off for a few hours. |
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| 01:07.15 | Micc | anyone know how to setup line keys that work on diffrent sip accounts on a mitel 5324, it keep saying it has to be the primary account when its line. |
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| 02:19.04 | p0t | Whats up guys? Ive been working on this vicidial server for the last 7 hrs straight and still no luck getting it to work. It is my first install. I had some questions im sure wouldnt take too much time does anyone have a minute? |
| 02:19.14 | p0t | it would be greatly appreciated |
| 02:19.30 | WIMPy | ~ask |
| 02:19.30 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 02:19.54 | WIMPy | And remember this is not #vicidial. |
| 02:20.47 | *** part/#asterisk deo_ (~deo@112.198.90.222) |
| 02:21.07 | p0t | ~ask |
| 02:21.08 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 02:21.15 | p0t | ~ask help |
| 02:21.15 | infobot | Nope, p0t! I won't ask "help" |
| 02:22.04 | p0t | Well I think the problem is on the asterisk side Not sure. Im getting " Im sorry that is an invalid extension" error msg |
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| 02:22.21 | p0t | #vicidial is empty i figured maybe you guys could help |
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| 02:24.20 | p0t | IDK if I need a sip trunk carrier from a third party or its supposed working using my normal internet connection. If I do need a 3rd party trunk carrier ill go do that. But if i dont I dont understand where to start to get that too work. any help would be greatly appreciated at this point. thanks ! |
| 02:24.54 | WIMPy | What do you want to do? |
| 02:26.58 | p0t | I want to run the predictive dialer for one user |
| 02:27.10 | p0t | I was trying to do a manual dial first though |
| 02:27.21 | p0t | jsut to test it and thats when i get the extension error |
| 02:28.11 | WIMPy | If you want to call phone number you need some way to connect to the PSTN. |
| 02:28.56 | WIMPy | Either via some hardware that interfaces to phone lines or via an ITSP. |
| 02:28.59 | WIMPy | ~itsp |
| 02:28.59 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 02:29.15 | WIMPy | Or you use a gateqay which is your own itsp in a box. |
| 02:29.44 | WIMPy | gateway |
| 02:32.58 | p0t | WIMPy: ok makes sense. When yous say my own gateway does that mean it has to be a seperate box? |
| 02:33.30 | WIMPy | That was the 3rd option, somehow inbetween the other two. |
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| 02:39.01 | p0t | Its making more sense |
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| 02:43.53 | p0t | WIMPy: How would I go about creating a sip gateway? |
| 02:44.32 | p0t | is there a way to have it on the same box ask the asterisk/vici server? |
| 02:44.53 | p0t | as* |
| 02:55.01 | WIMPy | You but it. |
| 02:55.06 | WIMPy | You buy it. |
| 02:55.28 | WIMPy | And yes, Asterisk can be used as a gateway as well. |
| 02:55.59 | WIMPy | That means you have to stuff some hardware in to connect to some sort of phone line. |
| 03:02.06 | p0t | umm ok |
| 03:02.40 | p0t | I was preffering the Free options. Thanks WIMPy big help!!!! |
| 03:18.09 | SeRi | Penguin: any use of the headset? |
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| 03:23.10 | p0t | can anyone recommend a cheap sip gateway |
| 03:24.46 | WIMPy | For what kind of line? |
| 03:25.17 | WIMPy | And how many of them? |
| 03:26.22 | p0t | I just need one station working now. Im not sure what kind of line im on. |
| 03:27.01 | p0t | Currently im set up like this Asterisk> lan> WWW |
| 03:27.05 | WIMPy | Station seems to be the opposite of what we've talked about so far. |
| 03:27.08 | p0t | Im trying to call local homes |
| 03:27.38 | WIMPy | That's back to square 1. |
| 03:28.02 | WIMPy | I guess you should sign up with an ITSP to do your testing. |
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| 03:29.01 | p0t | Im logging in to the server from a laptop using a softphone |
| 03:31.39 | Micc | Anyone know where I can get R8 mitel firmware? |
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| 03:42.26 | Micc | I really need to find the R8 or R9 firmware tonight. |
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| 08:06.23 | x1user | Is application SetCallerPres() still in asterisk 11? [Dec 18 03:05:49] WARNING[6111][C-000000ef]: pbx.c:4398 pbx_extension_helper: No application 'SetCallPres' for extension (spnet_incoming, 10194835924895612, 1) |
| 08:08.48 | x1user | Should be Set(CALLERPRES()=varialbe) |
| 08:10.18 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 08:10.49 | kaldemar | x1user: "This function is deprecated in favor of CALLERID(num-pres) and CALLERID(name-pres)." |
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| 08:28.32 | bombev | good morning |
| 08:28.50 | bombev | hi can you guys help me how to deal with large volume of asterisk logs |
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| 08:54.52 | x1user | I can see outgoing calls by "dialplan show extension@context", can i see in which context an incoming call would fall somehow? |
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| 09:00.10 | kaldemar | that does not show any calls, it shows an extension in your dialplan. |
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| 09:01.07 | davlefou | Bonjour, café and croissant for all. |
| 09:01.15 | kaldemar | x1user: the context that is used is defined in the channel configuration files. |
| 09:01.53 | x1user | I need to trace exactly how a numbers goes trough the dialplan. |
| 09:02.49 | kaldemar | use dialplan show and use the context you have defined for a peer/channel/x. |
| 09:07.53 | bombev | kaldemar is it safe if I delete those files in var/log/asterisk full full1 full2 full3.... |
| 09:09.16 | kaldemar | bombev: it won't harm asterisk in any way. |
| 09:10.54 | bombev | aha |
| 09:12.40 | bombev | after I delete those full full1.. |
| 09:12.46 | bombev | should I restart something or not |
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| 09:15.43 | kaldemar | bombev: if asterisk has rights to write in the dir, no. |
| 09:20.30 | bombev | thanks |
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| 10:17.14 | ghost75 | how would be minimum dial time to forward to mobile, i just want to hear it ring once and 1sec was too short (was not even ringing) |
| 10:18.40 | WIMPy | Wait till you get the message that it's ringing. |
| 10:19.57 | ghost75 | is there dial option for that? |
| 10:19.58 | kaldemar | ghost75: there is no constant time for it. it depends on the connection path from your system to the receiving device. and that can vary. a lot. |
| 10:20.20 | *** join/#asterisk ATS63 (dce94bd8@gateway/web/freenode/ip.220.233.75.216) |
| 10:23.59 | WIMPy | No Option, but you should get the message via AMI IIRC. |
| 10:24.39 | ghost75 | mmhh sounds like a lot code |
| 10:24.51 | ghost75 | i try with 5 seconds then |
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| 10:25.50 | WIMPy | As there's a radio link involved, you can be sure that the time will vary on each call. |
| 10:26.57 | WIMPy | But different operators might even make more of a difference. |
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| 10:57.10 | giany | hello |
| 10:57.24 | giany | anyone knows why asterisk would generate a "v17-12000-long-training" T38 reply? |
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| 11:26.43 | Diffen | Hello all, whats the best way to store information from an invite in a mysql db or just a json string or something similar. We want to send the incoming call information to our CRM system and then the customer card should pop-up when the call have reached the agent. Anyone that have any thougts on best practice? |
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| 11:29.34 | ATS63 | Diffen: if I were you, I'd check out System() |
| 11:29.47 | ATS63 | ie. execute a shell command |
| 11:29.59 | ATS63 | so you'd be more than able to write a script that does exactly what you want |
| 11:30.30 | Diffen | ATS63 ill check it out :) Thanks man |
| 11:31.17 | ATS63 | I like this example the most... |
| 11:31.33 | ATS63 | send a netbios msg with samba to a windows machine... |
| 11:31.35 | ATS63 | exten => 200,2,System(/bin/echo -e "'Incoming Call From: ${CALLERID} \\r Received: ${DATETIME}'"|/usr/bin/smbclient -M target_netbiosname) |
| 11:32.14 | Diffen | yes thats nice :). |
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| 11:35.33 | Diffen | hmm i belive i need to have some sort of list of that netbiosname are connected to caller id. So the right machine are called. There will be around 25 agents so i belive i need to have a check first. |
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| 11:38.02 | ATS63 | You'll most certainly have to make a script for your needs |
| 11:38.07 | ATS63 | That was just an example use |
| 11:38.12 | ATS63 | I don't even know if it works! |
| 11:38.57 | kaldemar | Diffen: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Database_id287624.html |
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| 11:45.14 | Diffen | Atleast i have something to read now so thanks :) |
| 11:58.01 | davlefouAMD | hi, in asteriskrt, where i put the register action? |
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| 13:22.32 | zamba | is it possible to get audio input from line feeded directly into one or several channels? |
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| 13:57.43 | Claies | good morning |
| 14:00.27 | Claies | anyone here good with sip connections that might give me a hand? I'm trying to figure out how to get my outgoing sip header from values to show something other than the station id |
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| 14:11.20 | leifmadsen | Claies: use the fromuser option to change them :) |
| 14:12.08 | Claies | I did, it is having no effect |
| 14:13.53 | Claies | I keep getting From: "8014" <sip:8014@97.76.29.142>; |
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| 14:17.38 | Claies | hmm what is the difference between fromuser and remote party id? |
| 14:21.27 | pabelanger | ~book |
| 14:21.27 | 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 |
| 14:21.33 | pabelanger | Claies: ^ |
| 14:21.34 | kaldemar | Claies: different headers for example. |
| 14:22.24 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:22.44 | Claies | I'm really actually upset with megapath, they have told me I'm sending the wrong info but can't or won't tell me how to send the correct info |
| 14:23.46 | jmetro | maybe they dont know what is wrong and are grasping at straws. |
| 14:24.34 | Claies | <PROTECTED> |
| 14:26.15 | *** join/#asterisk vassilux (~vassilux@LPuteaux-156-14-48-156.w82-127.abo.wanadoo.fr) |
| 14:30.46 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:30.46 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:36.35 | *** join/#asterisk TheCompWiz (~TheCompWi@63.214.236.169) |
| 14:43.25 | Katty | infobot: crittercam |
| 14:43.26 | infobot | [crittercam] The Birdie Breakfast Buffet! http://www.ustream.tv/channel/birdie-breakfast-buffet |
| 14:43.34 | Katty | squirrels have finally found the feeder ^____________^ |
| 14:44.06 | tzanger | lol |
| 14:44.14 | tzanger | are they as fat there as they are here at this time of year? |
| 14:44.24 | Katty | who knows. |
| 14:44.27 | tzanger | I swear some of them roll more than they scurry |
| 14:44.31 | Katty | they will be fat now that they've found the feeder tho. |
| 14:44.41 | Katty | i plan to spoil them rotten! |
| 14:44.42 | *** join/#asterisk wotanskrieger (c8c9a473@gateway/web/freenode/ip.200.201.164.115) |
| 14:44.42 | ghost75 | when i forward call to my cell phone, is it possible to show on display the original caller ? |
| 14:44.52 | tzanger | I. I used to be by the window, and I could watch the squirrels, and they were merry... |
| 14:44.53 | wotanskrieger | hi all |
| 14:45.25 | Katty | ghost75: yes (= |
| 14:45.53 | Katty | ghost75: what i did was make 2 digit mobile extensions. |
| 14:46.04 | Katty | ghost75: so for example dialing 45 would dial my cellphone |
| 14:46.34 | Katty | ghost75: rather than 7 or 10 or 11 digit, which would auto matically set our "main number" as the outbound callerid |
| 14:46.47 | Katty | ghost75: then, forward your phone to the 2 digit mobile extension, which does not tinker with the callerid. |
| 14:47.11 | Katty | ghost75: there are likely many ways to do it...that's just what i did. |
| 14:47.28 | ghost75 | my mobile dont use voip |
| 14:47.37 | Katty | neither does mine. |
| 14:47.47 | Katty | well that's not true ;> |
| 14:48.03 | Katty | it does. but voip doesn't really have anything to do with the above scenario. |
| 14:48.18 | wotanskrieger | anyone ever used a softphone with web interface? I want to install a softphone in my asterisk server which accessing an URL in a browser I can manager all its features. |
| 14:48.43 | Katty | wotanskrieger: that's a neat idea. |
| 14:48.45 | jmetro | sounds nifty. |
| 14:48.59 | Katty | wotanskrieger: i'd keep it internal if i were you ;> |
| 14:49.17 | jmetro | what you said suggested an * management UI not a softphone UI though |
| 14:49.35 | Katty | jmetro: why not both?! |
| 14:49.37 | jmetro | unless you mean "all its features" for the softphone. |
| 14:49.41 | jmetro | ^^^ exactly |
| 14:49.55 | wotanskrieger | Katty: Indeed. :) I found twinkle but I don't know if it's ideal to my solution. Check it out: http://www.twinklephone.com/ |
| 14:50.28 | jmetro | that looks like a normal softphone, not web based. |
| 14:50.31 | Robotman321 | hmm, there was a session at astricon that outlined a web based softphone, of sorts.. trying to remember which one it was.. Although you had to build it yourself ^^' |
| 14:51.03 | jmetro | Isymphony is very close but in the end it downloads a client. |
| 14:51.13 | Katty | i was just thinking that |
| 14:51.15 | ghost75 | i dont understand why this should display nr of original caller |
| 14:51.18 | Katty | it does look similiar to isymphony. |
| 14:51.44 | Katty | ghost75: why don't you look at the cli while it's forwarding |
| 14:51.48 | Robotman321 | Ah WebRTC allows for something like that.. but again, you have to do ityourself.. |
| 14:51.51 | Katty | ghost75: maybe it will tell you. |
| 14:52.04 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
| 14:52.35 | jmetro | by "build it yourself" do you mean configuration and compiling like with *, or do you mean "code the whole thing but heres a cool idea!" |
| 14:52.56 | Robotman321 | basically coding it yourself, using the WebRTC protocols.. |
| 14:52.58 | Katty | oh i imagine there'd be some templates |
| 14:53.07 | Katty | but you'd probably have to build the bulk of it yourself |
| 14:53.17 | wotanskrieger | jmetro: It's not a normal softphone, I think. It's a web-based UI. For example, Monast allow us to monitor asterisk extensions, doesn't it? So, a web-based softphone gives me advanced features to manager my server. |
| 14:53.21 | Katty | i don't do things like that very well. |
| 14:53.58 | wotanskrieger | jmetro: looking powerpoint twinkle presentation I can read this: Clicking a SIP URL in a web browser instructs Twinkle to make a phone call. |
| 14:54.00 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.82.33) |
| 14:54.19 | jmetro | Wotanskrieger - Correct, but softphones can normally dial SIP URI's |
| 14:54.33 | jmetro | that does not make it web based, your browser is calling Twinkle and twinkle associated itself. |
| 14:54.45 | wotanskrieger | jmetro: got it |
| 14:55.26 | wotanskrieger | jmetro: so... let's start from scratch again :P |
| 14:55.35 | jmetro | "web based" to me means you dont do an install on the client, like... like accessing webmail on Hotmail. |
| 14:56.14 | wotanskrieger | jmetro: yeah, it's my meaning too. I need to improve my english skills :P |
| 15:01.50 | *** join/#asterisk frek818 (~Adium@124.6.136.142) |
| 15:03.10 | wotanskrieger | jmetro: http://phono.com/ |
| 15:06.38 | jmetro | Now that looks like a browser softphone |
| 15:06.59 | leifmadsen | Not sure if anyone here is in the Ottawa, ON area, but I'm hiring for a Junior UC Lab/QA Technician. If you or someone you know might be interested, please msg me directly. |
| 15:11.46 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 15:12.24 | jmetro | Wotanskrieger: That unfortunately looks like it's limited to dialing through the Phono company to the Voxeo cloud though... |
| 15:13.09 | wotanskrieger | jmetro: I found 2 forums doing reference to this one: http://www.mizu-voip.com/Products.aspx |
| 15:16.02 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 15:16.13 | *** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 15:16.16 | wotanskrieger | jmetro: http://blog.svnlabs.com/sip-web-phone/ |
| 15:17.18 | wotanskrieger | jmetro: well, sorry if I'm boring you with these links. I'll try to find an ideal solution. Thanks :) |
| 15:17.22 | *** join/#asterisk passerby (~a.kozhukh@193.109.166.252) |
| 15:17.43 | *** part/#asterisk passerby (~a.kozhukh@193.109.166.252) |
| 15:25.26 | jmetro | I like the sipwebphone one, actually, but its paid-for |
| 15:26.42 | *** join/#asterisk hesco (~hesco@174.48.250.91) |
| 15:30.39 | hesco | On a vanilla asterisk installation, I added this line to my dialplan: |
| 15:30.40 | hesco | exten => _404thisdid,n,Set(__CALLERID(name)=prefix:${CALLERID(name)}) |
| 15:30.40 | hesco | My tester reports: 'same story' whether I quote the prefix or not. How do I prefix my callerID? |
| 15:31.58 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:31.59 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:32.20 | hesco | by tester, I mean call recipient and by 'same story' I mean name: unknown, and my originating number, w/o a prefix, although I also added my prefix to the CALLERID(num) as well |
| 15:32.39 | *** join/#asterisk SuPrSluG (~SuPrSluG@rrcs-50-75-185-122.nys.biz.rr.com) |
| 15:37.44 | hesco | FYI, I used asterisk -rx "dialplan reload" after every change and before each test |
| 15:37.54 | *** join/#asterisk TheCompWiz (~TheCompWi@63.214.236.169) |
| 15:43.49 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:46.58 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:46.58 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:47.00 | bombev | have a good one |
| 15:47.46 | cusco | I'm still having calls dropped at 32 seconds, and I can't figure out why... http://paste.debian.net/217020/ call starts in line 42 |
| 15:50.42 | Katty | knits gloves. |
| 15:52.32 | cusco | :| |
| 15:54.21 | *** join/#asterisk PipBoy (~PipBoy@66.212.187.33.tor.pathcom.com) |
| 15:55.36 | [TK]D-Fender | <hesco> My tester reports: 'same story' whether I quote the prefix or not. How do I prefix my callerID? <- you shove chracters in front of the value you assign the function. |
| 15:56.07 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 15:59.12 | PipBoy | Anyone have experience limmiting a Queue agent to 2 calls. seems like the only option is "1 call makes an agent busy, and then you can skip that agent" |
| 16:01.29 | jmetro | Autofill seems to be an option maybe. |
| 16:02.01 | jmetro | http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf |
| 16:02.42 | PipBoy | Thank you, I will look into it |
| 16:03.18 | PipBoy | I could also make a phone have two extensions on each line key and they can each handle only one call.. but that feels kinda dirty |
| 16:03.49 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 16:03.49 | jmetro | right |
| 16:04.00 | jmetro | i dont think that would...work the way you want. |
| 16:04.26 | jmetro | autofill gives one person all the calls and they route, but if youre looking for 1+1 on each member, that might be different. |
| 16:04.29 | cusco | [TK]D-Fender: perhaps you cold take a look, i'm sure it has the whole call now http://paste.debian.net/217020/ |
| 16:05.55 | [TK]D-Fender | cusco, Not at all, just las yesterday |
| 16:05.59 | [TK]D-Fender | like* |
| 16:06.41 | cusco | [TK]D-Fender: it must, I echoed a string before dialing to full log, and echoed another after hanging up |
| 16:06.57 | cusco | and got all the log between those strings |
| 16:07.29 | [TK]D-Fender | cusco, And you missed the point. You may have from the TIME of the beginning and the end, but its the content in the middle that is lacking. |
| 16:07.38 | [TK]D-Fender | CunningPike, You are not looking at the complete call. |
| 16:07.42 | [TK]D-Fender | cusco, ^ |
| 16:07.58 | cusco | [TK]D-Fender: I did not filter anything else |
| 16:08.10 | cusco | except replacing the mysql connection user/pass |
| 16:08.19 | cusco | verbose is set to 15 |
| 16:08.27 | cusco | sip debug is set do my peer and destination peer |
| 16:08.30 | cusco | what am I missing? |
| 16:08.48 | [TK]D-Fender | <cusco> sip debug is set do my peer and destination peer <- NOT POSSIBLE, and this is the screwup |
| 16:09.00 | cusco | not possible? |
| 16:09.05 | cusco | sip set debug peer 150 |
| 16:09.05 | [TK]D-Fender | cusco, If you actually looked at what you PB'd you should see that instantly. |
| 16:09.13 | cusco | o.O |
| 16:09.22 | cusco | I am looking at it |
| 16:09.29 | cusco | I am not understanding what you mean |
| 16:09.42 | [TK]D-Fender | cusco, What do you see SIP debug to/from in there? |
| 16:10.05 | cusco | from 150 to 0404 |
| 16:10.14 | cusco | er |
| 16:10.16 | cusco | 404 |
| 16:10.17 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 16:10.38 | cusco | I could enable the whole sip debug, but it will be harder to read |
| 16:10.46 | cusco | would you preffer that way? |
| 16:10.53 | [TK]D-Fender | cusco, No, you don't. First, those 2 devices do not directly talk to each other. |
| 16:11.10 | cusco | [TK]D-Fender: its a local channel, in the local dialplan it dials SIP/404 |
| 16:11.29 | [TK]D-Fender | <[TK]D-Fender> cusco, What do you see SIP debug to/from in there? |
| 16:11.45 | cusco | ok I saw it now |
| 16:11.59 | cusco | only one end |
| 16:12.30 | cusco | ok hold on |
| 16:16.53 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 16:22.55 | hesco | [TK]D-Fender: you advise: "you shove chracters in front of the value you assign the function.", but how is this not doing exactly that? exten => _404thisdid,n,Set(__CALLERID(name)=prefix:${CALLERID(name)}) |
| 16:23.50 | hesco | I've tested w/ zero, one or two underscores prefixing the variable name. I've tested the prefix w/ and w/o quotes. |
| 16:24.07 | cusco | [TK]D-Fender: I can't find a pastebin that works with this large text but the paste is here: http://62.28.187.252/geada/call.txt -- now it has the whole call and the whole sip. I am peer 150, and destination is peer 404 (with ip 10.10.10.39) |
| 16:24.08 | hesco | nothing seems to work, what is wrong with this syntax? |
| 16:24.46 | [TK]D-Fender | hesco, Who told you to put junk in front of your function in that Set? |
| 16:25.18 | [TK]D-Fender | <[TK]D-Fender> <hesco> My tester reports: 'same story' whether I quote the prefix or not. How do I prefix my callerID? <- you shove chracters in front of the value you assign the function |
| 16:26.24 | hesco | The *book says doing so will permit that value to be inherited. |
| 16:26.31 | cusco | hesco: I wanted to set a cdr(var) too, and I had to set it in each part of the dialplan,, however I did Set(__UID=${UNIQUEID}) in the beginning of the dialplan, and used Set(CDR(uid)=${UID}) in every other step of the call (where it would create a new channel) |
| 16:26.44 | cusco | hesco: that applies to a variable, not a function |
| 16:26.50 | cusco | (if I am not mistaken) |
| 16:26.59 | *** join/#asterisk wonderworld (~w@77.181.56.216) |
| 16:27.37 | hesco | so which is CALLERID? a variable ro a function? I assumed it to be an array with two named keys, num and name |
| 16:27.46 | [TK]D-Fender | hesco, a VARIABLE. This has nothing to do with CALLERID |
| 16:28.03 | [TK]D-Fender | "core show function CALLERID" <- what do you think? |
| 16:28.23 | [TK]D-Fender | There may be a hint in there ;) |
| 16:29.19 | hesco | looks like a function when you put it that way |
| 16:29.36 | hesco | I had imagined a channel variable. thanks for clarity |
| 16:31.00 | cusco | tk I see a bye, and then a reply 200 ok from me (150) before the hanging up |
| 16:31.32 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.38) |
| 16:31.40 | cusco | and I am guessing that asterisk places that BYE |
| 16:34.28 | *** join/#asterisk rbowles (~bb@75-147-63-201-NewEngland.hfc.comcastbusiness.net) |
| 16:36.07 | *** join/#asterisk TheCompWiz (~TheCompWi@63.214.236.169) |
| 16:39.21 | cusco | the BYE originates from the called peer |
| 16:39.47 | *** join/#asterisk lorsungcu (~anonymous@67.136.169.134) |
| 16:40.00 | cusco | but I still can't figure why |
| 16:43.45 | hesco | dialplan now includes: exten => _404thisdid,n,Set(CALLERID(num,pre:${CALLERID(num)})) |
| 16:43.45 | hesco | and *CLI> reports: "CALLERID(num,pre:404myphone)"), but my call recipient reports a number, but no prefix. |
| 16:44.18 | [TK]D-Fender | hesco, that is not how you set a function.... |
| 16:45.30 | cusco | hesco: Set(CALLERID(num)=${prefix}${CALLERID(num)}); I would guess |
| 16:47.20 | hesco | cusco: I assume your context assumes that prefix has previously been assigned, right? here I am using 'prefix:' as a string. |
| 16:48.24 | [TK]D-Fender | hesco, Your last failed example showed us a Set() with an EQUALS SIGN. You should revisit chapter 5 of THE BOOK |
| 16:48.25 | [TK]D-Fender | ~book |
| 16:48.26 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:49.12 | [TK]D-Fender | without* |
| 16:54.38 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.6.188) |
| 16:56.44 | Penguin | Contexts don't set variables. |
| 17:01.14 | hesco | OK, now using: Set(CALLERID(num)=pre:${CALLERID(num)}); and a NoOp(Our prefixed CID is now: ${CALLERID(num)}) suggests I got this right, although the recipient phones do not seem to display the prefix, only the number. |
| 17:01.40 | Penguin | Is the prefix a number? |
| 17:01.51 | hesco | no, alpha |
| 17:02.01 | Penguin | Try changing it to a number and see if it displays that. |
| 17:02.10 | hesco | in this case: 'pre:' |
| 17:02.34 | Penguin | I bet you're going to find out that callerid number means NUMBER, not letter and special character. |
| 17:02.38 | hesco | interesting test, but not my use case. |
| 17:03.05 | hesco | its not displaying the (name) at all with my test phone. |
| 17:03.28 | hesco | out of time for this this morning, though. Thanks. |
| 17:03.57 | kaldemar | hesco: you'll be better off modifying the name part. that way your users can call the number back too. |
| 17:05.56 | hesco | this test came up an unprefixed, unkown cid, when I tried to set CALLERID(name) |
| 17:06.30 | hesco | Unknown No number when inspecting call history. |
| 17:06.38 | TheCompWiz | weeeeeeeeee... |
| 17:06.55 | hesco | I assume that this partly depends on how each receiving device implements things as well. |
| 17:09.31 | hesco | prefixing name, leaving number alone gives me Unknown, valid phone number. And with that test I am really running to the office. |
| 17:09.40 | hesco | Thanks for help folks. |
| 17:10.12 | *** join/#asterisk felipealmeida (~user@querubim.tecgraf.puc-rio.br) |
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| 17:27.23 | zamba | is it possible to get live audio from a sound card fed into a channel? |
| 17:28.10 | *** join/#asterisk TheCompWiz (~TheCompWi@63.214.236.169) |
| 17:35.43 | rrittgarn | What update message gets sent to a carrier when you transfer to an extension? I'm getting a server 500 when i transfer from an IVR and the provider is saying that its an unsupported Update Message. |
| 17:36.38 | rrittgarn | capturing all traffic coming out and having difficulty finding the 1 packet that seems to be causing my issue... Symptom is once i press the extension in the IVR audio becomes one way (I don't get any back from Level3) |
| 17:38.17 | *** join/#asterisk navaismo (~Administr@189.191.20.105) |
| 17:46.16 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:02.04 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 18:02.46 | Katty | hello my asterisk does not work at all how to fix plz?? urgent thx |
| 18:03.11 | *** join/#asterisk ageis (kevin@ageispolis.net) |
| 18:06.46 | *** join/#asterisk prg3 (~prg3@chatter.majestik.org) |
| 18:19.44 | Nugget | are you loading chan_huggles? |
| 18:20.07 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.174) |
| 18:24.15 | *** join/#asterisk frek818 (~Adium@124.6.136.142) |
| 18:24.21 | [TK]D-Fender | It was deprecated in lieu of chan_pamples |
| 18:28.13 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 18:38.08 | *** join/#asterisk corretico (~luis@190.211.93.38) |
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| 18:40.43 | sruffell | chuckles |
| 18:45.44 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-81-244.user.veloxzone.com.br) |
| 18:48.18 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.127) |
| 18:48.22 | Katty | Nugget: ! |
| 18:48.25 | Katty | glomps Nugget |
| 18:48.37 | Nugget | eek |
| 18:49.01 | Katty | how'rechu |
| 18:51.22 | anonymouz666 | very strange DTMF situation when using Linksys PAP2 and Asterisk 1.8.16... it triggers the transfer (feature code) but after that, DTMFs are rarely recognized... always missings digits. |
| 18:51.32 | anonymouz666 | DTMF mode 2833 |
| 18:51.48 | Katty | you wanna know the craziest dtmf issue i've ever seen? |
| 18:52.05 | anonymouz666 | sure |
| 18:52.05 | Katty | a client of mine used to use a radio as their on hold music |
| 18:52.16 | Katty | and one of the morning shows featured phone calls. |
| 18:52.30 | Nugget | hung over. |
| 18:52.34 | Katty | and the phone system would pick up the user that was on hold as dialing an extension |
| 18:52.39 | Nugget | Too much scotch last night and I'm all foggy-headed today |
| 18:52.44 | Katty | when it was really the on hold music |
| 18:52.48 | Katty | course that was on a toshiba box. |
| 18:52.55 | Katty | pats Nugget's head |
| 18:53.08 | Katty | time for water and tylenol :< |
| 18:53.13 | anonymouz666 | hehe |
| 18:55.48 | Katty | don't have anything on my blog about dtmf issues. |
| 18:55.58 | Katty | :< |
| 18:56.03 | anonymouz666 | the feature code always work (#1) |
| 18:56.14 | anonymouz666 | only the dst number does not |
| 18:56.19 | anonymouz666 | this is bizarre |
| 18:57.09 | Katty | what happens if you read the dtmf and play it back |
| 18:57.13 | Katty | does it spaz |
| 19:03.57 | rrittgarn | Anyone know how to tell the PBX to not send an update for RPID http://pastebin.com/QB8W0yi3 |
| 19:04.38 | Qwell | disable it? |
| 19:14.59 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 19:22.06 | leifmadsen | rrittgarn: sendrpid=no ? |
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| 19:25.28 | capnkooc | Katty, what is the url of your blog? |
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| 19:41.00 | Katty | hugs Qwell |
| 19:41.06 | Qwell | O.o |
| 19:41.08 | Katty | capnkooc: it's private. |
| 19:41.39 | capnkooc | Katty, oh.. ok |
| 19:43.26 | Katty | prods Qwell with a knitting needle. |
| 19:43.40 | Qwell | EEP |
| 19:43.44 | Katty | :> |
| 19:43.51 | carrar | We should outlaw those tools |
| 19:44.11 | Katty | :< |
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| 19:48.54 | jmetro | only assault needles |
| 19:49.25 | file | o.o |
| 19:49.56 | Katty | hi file |
| 19:50.11 | anonymouz666 | Katty: inband set solved the issue |
| 19:50.19 | file | hola |
| 19:50.25 | anonymouz666 | something is broken with rfc2833 |
| 19:50.30 | Katty | :> |
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| 20:10.03 | rbowles | rfc2833 is not an 'exact science' |
| 20:10.36 | rbowles | i find huge variations depending on SIP clients on what they actually send |
| 20:11.25 | rbowles | each one probably tweak to deal with some odd gateway's idea of what it should be |
| 20:11.29 | Maliuta | It's not the RFC that is broken ... it's the implementations. |
| 20:11.42 | Maliuta | It happens with most RFC's |
| 20:13.05 | Maliuta | Even the ISC admits to having not implemented things in and RFC compliant way. The worst is when it comes to SMTP, a large amount of code in some of the better MTA's is to detect and deal with broken implementations. |
| 20:13.18 | rbowles | agreed! |
| 20:14.08 | Maliuta | I blame dev's ... but them I'm a sysadmin and they make my life hell, so I blame them for the bad weather ;) |
| 20:14.33 | Maliuta | s/them I'm/then I'm/ |
| 20:15.38 | rbowles | I suspect, however, that SIP clients are tweaked to satisfy what some gateways need to work, so if a widely used gateway isn't "forgiving about what it accepts", the SIP clients might get tweaked to accomodate |
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| 20:21.11 | Maliuta | almost every client I have tried works with *, and that's about all I need |
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| 20:38.11 | TheCompWiz | weeeeeeeeeeeee |
| 20:41.43 | rbowles | i think asterisk is 'forgiving about what it accepts' ... our upstream DID providers and downstream termination providers ... not so much |
| 20:42.24 | rbowles | i've had similar results with asterisk IVR scripts working nicely with any SIP client and rfc2833 |
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| 23:31.09 | ideaman55 | Alright guys, I'm doing an IAX2 to IAX2 connection with g723.1, and I have the TC400 card here, and 20 licenses of the software codec on my gateway that is the other IAX peer. That GW dumps my calls out a sip provider which I have set to disallow all and allow=ulaw. My iax2 config is set disallow=all and allow=g723.1, but I get the message saying it had to drop the call because it couldn't |
| 23:31.15 | ideaman55 | make the iax channel compatible with the sip channel. Thoughts? |
| 23:31.44 | ideaman55 | g729 works in it's place in iax though |
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