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| 05:21.32 | ruben23 | hi guys on a recording filename of asterisk...i see a variable EPOCH...what this mean..? any idea.>? |
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| 05:30.37 | Penguin | You've been using Unix-like operating systems for how long now, and you don't know what epoch means? |
| 05:34.42 | ruben23 | <PROTECTED> |
| 05:34.50 | ruben23 | what does it mean..? |
| 05:35.13 | Penguin | What does google say "epoch" means? |
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| 05:38.13 | dkakoti | HI all how can I add sipaddheader in call file? |
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| 05:57.20 | ChannelZ | I'm not sure you can.. |
| 05:57.28 | dkakoti | OK |
| 05:58.54 | ChannelZ | Probably best to write your own dialer as AMI/ARI |
| 06:00.04 | dkakoti | Ok.Which ll be good?AMI or ARI? In case of AMI I am concern for security. |
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| 06:01.16 | ChannelZ | well depends on your version partially, ARI being new in v12. |
| 06:01.37 | dkakoti | I am using asterisk 11. |
| 06:01.55 | ChannelZ | so AMI it is then :) |
| 06:02.18 | dkakoti | k thanks @Channelz |
| 06:02.54 | ChannelZ | as for security, if it's a shared environment, make sure you pick a strong PW and then make manager.conf only readable by asterisk so someone else couldn't look at it |
| 06:03.19 | dkakoti | ok |
| 06:03.24 | ChannelZ | if you're worried about external accessability, it can listen only to localhost |
| 06:04.17 | Penguin | I would create a bit of dialplan using the application you want and then call a local channel from the call file. |
| 06:04.41 | dkakoti | yes u r right . I can use the local asterisk machine to dial either using PRI or I can create trunk and dial using that. |
| 06:06.07 | ChannelZ | yeah guess you could do what Penguin said - you'd just have to pass in the number you were ultimately wanting to call from your script or whatever would be making the call file |
| 06:06.49 | fling | Which firmware to use for e1550 voice support? |
| 06:07.01 | fling | e1553 does not flash giving an 'error' |
| 06:07.12 | dkakoti | @Penguin Yes I can do that. |
| 06:07.46 | dkakoti | I think AMI ll be straight forward. |
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| 06:56.45 | lkthomas | hey guys |
| 06:57.02 | lkthomas | does normal ATA be able to convert incoming call to VoIP ? |
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| 07:03.46 | Penguin | no |
| 07:04.18 | Penguin | To connect an ATA to the phone line from the wall jack, you need a device with FXO port on it. Regular ATA has FXS ports. |
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| 07:44.51 | lkthomas | Penguin: okay, thanks |
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| 08:26.53 | fling | Are there any changes needed to my sip.conf and extensions.conf when upgrading from 1.8 to 11? |
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| 08:46.35 | Milos | fling, I know nat= changed. but that's all I know for now. |
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| 08:47.13 | Milos | e.g. nat=yes now nat=comedia,force_rport I believe |
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| 08:55.27 | wdoekes | fling: there is a CHANGES file you should look at |
| 08:56.02 | wdoekes | (and the UPGRADE file, to get to know new features) |
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| 09:13.34 | madax | hi guys, i need a confirmation about the permissions in the manager.conf |
| 09:14.10 | madax | is read=all write=all is enough for example to use the method originate via the ami |
| 09:14.24 | madax | or should i add the originate perm as well ? |
| 09:14.51 | madax | and the system like write=all,system,originate ? |
| 09:15.28 | madax | i am using asterisk 11 |
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| 09:51.11 | fling | Milos: I'm not using nat but I will try using it soon ;> |
| 09:51.20 | Milos | :) |
| 09:51.28 | fling | Milos: I had nat=comedia,force_rport on some old 1.8 server btw |
| 09:51.33 | fling | wdoekes: thanks :> |
| 09:51.50 | fling | I'm about to install into an lxc container and I'm thinking about not using nat somehow |
| 09:51.51 | Milos | I guess nat=yes was removed then because it was stupid |
| 09:53.42 | Penguin | nat=yes is the same as nat=comedia,force_rport |
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| 10:07.12 | Milos | that's what I said |
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| 10:09.04 | Penguin | But nat=yes MEANS nat=comedia,force_rport |
| 10:14.08 | anonymouz666 | Penguin: so what happens if there are two media servers in comedia mode, each one expecting to receive the first audio packet? |
| 10:14.13 | anonymouz666 | media deadlock |
| 10:14.15 | anonymouz666 | he he |
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| 10:22.21 | fling | What is the proper way to play an audio file to a channel? |
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| 10:31.29 | fling | How to automagically drop old calls? I have a lot of things in 'show channels' |
| 10:31.35 | fling | rtp timeout? |
| 10:33.19 | anonymouz666 | fling: make sure you won't drop active calls, 'core show channels' shows a a lot of things when you have calls :) |
| 10:34.46 | fling | anonymouz666: do I need rtptimeout or what? |
| 10:35.09 | fling | anonymouz666: for example there are channels left after sip client crash. |
| 10:35.19 | fling | anonymouz666: or from a bot spam call. |
| 10:35.38 | anonymouz666 | rtptimeout could help channels that uses RTP, yes |
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| 11:06.05 | Milos | Penguin, I read that net=yes is deprecated/removed |
| 11:06.07 | Milos | s/net/nat/ |
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| 11:06.27 | Penguin | It's deprecated, but should not yet be removed. |
| 11:06.32 | Milos | ah ok |
| 11:06.47 | Milos | well I told him that in upgrading > 1.8 he should replace nat=yes with the other variant |
| 11:06.48 | Milos | that's all |
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| 11:24.38 | dan_j | Hi. When using gototime, if I want do 8am till midnight, am i correct that i do 08:00-00:00? Or do i have to do 08:00-23:59:59? |
| 11:26.34 | dan_j | *if i want TO do" :-S |
| 11:27.56 | Penguin | 08:00-00:00 would be 8am through midnight |
| 11:28.10 | Penguin | I don't think you can specify seconds anyway. |
| 11:28.20 | dan_j | Ok. Thanks. |
| 11:29.12 | dan_j | Also if I want to specify something like Monday 8am through to Tuesday 10am, do i need to use multiple GotoIfTime commands? |
| 11:37.44 | Penguin | I think you'd need one for monday 8am to midnight and then tuesday midnight to 10am. |
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| 11:43.17 | nfi|ermes | hi all |
| 11:43.20 | nfi|ermes | [Nov 10 12:36:07] NOTICE[18348] chan_sip.c: Call from '' (212.129.8.166:5070) to extension '000972595237548' rejected because extension not found in context 'default'. |
| 11:43.47 | dan_j | Penguin: perfect. thought so. thank you |
| 11:44.02 | nfi|ermes | this means that someone from 212.129.8.166 succeded to register/login in asterisk ??? |
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| 11:48.04 | nfi|ermes | how can anyone try to make a call whithouth being registered ? in case he is registered how can i know which extension is using ( Call from '' ) ? |
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| 12:00.21 | WIMPy | nfi|ermes: Registration is only required to receive calls (unless an IP is configured). It's in no way related to making calls. |
| 12:03.06 | anonymouz666 | nfi|ermes, INVITE is one thing, REGISTER another. Some phones your have an option to make calls only if registered. |
| 12:03.18 | anonymouz666 | you can even AUTH your INVITE without being registered. |
| 12:03.29 | anonymouz666 | SIP is very flexible protocol |
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| 12:11.30 | anonymouz666 | .se has joined |
| 12:12.09 | anonymouz666 | .se is definitely a country a should visit |
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| 12:21.36 | dan_j | Penguin: Following on from my last question, do you know if '14 Nov 00:00 to 00:00' would be 24 hours? |
| 12:22.47 | WIMPy | dan_j: If you have more complicated patterns, I'd consider changing an AstBD entry via cron instead. |
| 12:30.45 | anonymouz666 | complicated patterns is a bad thing :) |
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| 12:55.31 | marceloamorim | guys, there is any parameters that helps a TDM410p identify an answer of a call? |
| 12:57.49 | anonymouz666 | it's analog, you must answer it. |
| 12:57.56 | anonymouz666 | Answer() |
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| 12:58.11 | anonymouz666 | you mean when you dial? |
| 12:58.11 | marceloamorim | I mean, the other side, when I call for some body |
| 12:58.41 | anonymouz666 | are you using callprogress? |
| 12:58.59 | marceloamorim | should I? |
| 12:59.11 | anonymouz666 | if yes, disable it. DAHDI will answer the channel as soon as you dial. |
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| 13:00.29 | anonymouz666 | analog lines is a pain. :) |
| 13:00.37 | marceloamorim | ok, this option was enabled |
| 13:01.28 | marceloamorim | I`ll remove |
| 13:01.31 | marceloamorim | thx anonymouz666 |
| 13:02.22 | anonymouz666 | marceloamorim: it's good to help someone from the same country. |
| 13:03.39 | marceloamorim | yeah =( |
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| 13:52.01 | anonymouz666 | I am just impressed about how good videos exists in asterisk official youtube channel. |
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| 15:33.31 | dan_j | Hi. Here's an interesting one. A client needs to contact a number of large companies but the queue lengths are massive and he's wasting money paying his staff to listen to music. |
| 15:34.09 | dan_j | Is there any way to detect when a call has been answered by a human after the call has been through the queue? |
| 15:34.21 | WIMPy | Not really. |
| 15:34.24 | [TK]D-Fender | No |
| 15:34.35 | [TK]D-Fender | And no reasonable way to expect this to exist |
| 15:34.42 | WIMPy | You can try the "reverse hold" idea. In the hope the called party plays along. |
| 15:35.03 | [TK]D-Fender | Queues very oftern have human recordings with information that would be taken as being human. |
| 15:36.07 | dan_j | I said that to the client. Was just wondering if anyone has seen an experimental solution. A system that learns the sounds that recorded queues generate and triggers when the sound changes from the norm. |
| 15:36.59 | WIMPy | If you call the same queue over and over again, that should work. But I haven't heard of anyone trying. |
| 15:37.28 | WIMPy | But if it's the same destination(s), don't they offer some premium line without waitung? |
| 15:37.56 | [TK]D-Fender | [10:33]dan_jHi. Here's an interesting one. A client needs to contact a number of large companies but the queue lengths are massive and he's wasting money paying his staff to listen to music. <- I just continue working on other things while I'm on hold waiting... |
| 15:39.21 | WIMPy | Thinking of the whole thing.. There used to be queues that signalled the call being on hold while waiting. ... Those were the days... |
| 15:39.48 | dan_j | This particular company require you to start a new call for each case. and these staff are tasked with dealing with cases, so its a little difficult to do other things. |
| 15:40.26 | dan_j | I think i'll get them to install a second sip phone for each staff, so they can have call-hold on speakerphone, and continue their work on the other phone. |
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| 15:40.40 | WIMPy | BTW: Does chan_pjsip support hold in both directions? |
| 15:41.02 | dan_j | I did suggest paying someone minimum wage to make and route calls through to staff once answered. |
| 15:41.33 | WIMPy | That should work. |
| 15:44.16 | mjordan | WIMPy: chan_pjsip's hold behavior is basically the same as chan_sip. |
| 15:44.40 | WIMPy | So receive-only. |
| 15:44.57 | anonymouz666 | mjordan: looking forward for the Asterisk 13 webinar! |
| 15:45.24 | mjordan | WIMPy: today, there's no way to pass a hold through |
| 15:45.34 | mjordan | at the same time, there's no reason why that enhancement couldn't be made :-) |
| 15:45.37 | anonymouz666 | WIMPy: sorry I didn't ready the whole subject, but you want to forward the MOH instead generate one? |
| 15:45.59 | anonymouz666 | *read |
| 15:46.28 | mjordan | anonymouz666: thanks! Hope it lives up to the expectations :-) |
| 15:46.35 | WIMPy | mjordan: What to you mean by "pass through"? It is passed from chan_sip to other channels. Just not the other way round. |
| 15:46.54 | file | I added the enhancement |
| 15:47.02 | WIMPy | anonymouz666: No it's about signalling hold state. |
| 15:47.20 | file | in trunk you can specify that hold should be passed through in chan_pjsip so it will send an outgoing re-INVITE with the proper state |
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| 15:47.40 | WIMPy | file: Nice! |
| 15:49.05 | mjordan | file: heh, I thought you might have, but I couldn't find the code review :-P |
| 15:49.30 | file | it was in the past, we are in the future |
| 15:50.01 | mjordan | file: what is time? |
| 15:50.10 | file | mjordan, time is a concept of human creation |
| 15:50.11 | WIMPy | still thinks th future is over and we're back in the past. |
| 15:50.33 | file | or something |
| 15:50.52 | WIMPy | Does anyone know if time really exists? |
| 15:51.19 | file | we have no time to wonder about time! |
| 15:51.50 | WIMPy | Stop building deadlocks. |
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| 15:56.30 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 16:16.59 | *** join/#asterisk w3rt (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
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| 16:36.11 | *** join/#asterisk zacoben22 (5371477c@gateway/web/freenode/ip.83.113.71.124) |
| 16:36.47 | zacoben22 | Hi , is it possible in sip.conf to register different sip trunk ? |
| 16:37.13 | zacoben22 | I mean I have some sip to register from one domain and another one from another domain |
| 16:38.00 | zacoben22 | how can I switch the context depending on which Sip line is called ? |
| 16:41.22 | [TK]D-Fender | Don't use "register = >" to do your registrations. |
| 16:41.28 | WIMPy | Your question is a little vague. You can register as may accounts as you like and the peer definitions allow you to set a context for the calls. |
| 16:41.40 | [TK]D-Fender | There is a peer-based feature to use it's settings for the registration and trigger from there |
| 16:41.47 | [TK]D-Fender | Go read the sample confifgs |
| 16:42.03 | [TK]D-Fender | [11:38]zacoben22how can I switch the context depending on which Sip line is called ? <- that's what PEERS are for |
| 16:42.32 | WIMPy | Would have been nice if someone had added that to iax.conf as well, BTW. |
| 16:43.23 | mjordan | WIMPy: Patches welcome? :-D |
| 16:43.54 | WIMPy | I know. Unfortunately I can't do full-time Asterisk. |
| 16:44.06 | WIMPy | And I'm not even sure that would be enough for what I'd like to see. |
| 16:48.41 | zacoben22 | I have only one [general] section and I can set there only one context . so how can I do to make a certain context for a certain line , and another for the other ? |
| 16:49.06 | anonymouz666 | zacoben22: if your domains point to different IPs ok, use PEERS. If your domains point to the same IP, you have to use ONE peer and inspect some header to know what was dialed. |
| 16:49.22 | [TK]D-Fender | zacoben22: make a PEER to match the calls |
| 16:49.32 | WIMPy | zacoben22: I think you should start by reading the |
| 16:49.37 | WIMPy | ~book |
| 16:49.37 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:50.52 | zacoben22 | http://pastebin.com/cw58Z3zg |
| 16:51.54 | zacoben22 | here is the start of my sip.conf |
| 16:52.00 | anonymouz666 | zacoben22: if your read the book now, you will save a lot of time in the future... because you will understand how incoming calls are matched |
| 16:52.03 | zacoben22 | and I can set only one context |
| 16:52.37 | WIMPy | zacoben22: that's because you didn't read the documentation. |
| 16:52.56 | zacoben22 | I am quite new to asterisk and I wish that you helped me :) |
| 16:53.06 | WIMPy | You have already been told how to do it. |
| 16:53.46 | zacoben22 | if you knwo how to do it why don't you tell me how to do instead of me buying and reading hundreds of pages |
| 16:54.21 | WIMPy | You didn't even read the link, otherwise you'd know that you don't have to buy anything. |
| 16:54.33 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 16:54.44 | [TK]D-Fender | zacoben22: The book tells you how to make peers. Who is going to hand-type the exact same information into IRC when you're unwilling to read what's already published and available free to you? |
| 16:54.45 | WIMPy | But if you want someone to write your configs for you, then you do have to buy support from somewhere. |
| 16:55.15 | zacoben22 | where can I pay someone to do it ? |
| 16:55.52 | WIMPy | There's an asterisk-biz mailing list for that purpose. |
| 16:56.20 | WIMPy | Or you're lucky in here. |
| 16:57.00 | zacoben22 | you could help me in 3 minutes I am sure |
| 16:57.12 | zacoben22 | be nice with me :) |
| 16:58.28 | WIMPy | Maybe, but I'm already working on somethign else. Like most people. |
| 17:13.44 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.0.0 (2014/10/25), 11.14.0 (2014/11/10), 1.8.32.0 (2014/11/10); Standard: 12.7.0 (2014/11/10); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 17:16.31 | *** join/#asterisk danjenkins_ (~dan@nat/digium/x-btnzcmlolcvltxob) |
| 17:17.14 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 17:21.41 | *** join/#asterisk centurysam (~sanchez@OTWAON0916W-LP140-02-1279621625.dsl.bell.ca) |
| 17:22.04 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:da4:224e:7fff:fe5b:2df4) |
| 17:22.51 | centurysam | looking for help with basic setup that allows uses usernames for username instead of extensions |
| 17:23.14 | [TK]D-Fender | usernames are usernames |
| 17:23.24 | [TK]D-Fender | extensions are extensions |
| 17:23.33 | [TK]D-Fender | They have nothing to do with each other |
| 17:23.47 | centurysam | no no it works if i use extension as username rather then when setup with an actual username [jsmith1] vs [101] |
| 17:24.39 | [TK]D-Fender | if in sip.conf you have [jsmith1] then that us the username unless you set "username" or "defaultuser" as a parameter |
| 17:25.02 | centurysam | i did use [jsmith1] but then extensions start going completely screwy in extensions.conf |
| 17:25.10 | [TK]D-Fender | exten => 101,1,NoOp() <- This is an extension. This is something a device can dial if it is set to access the context you put it in |
| 17:25.13 | [TK]D-Fender | They have nothing to do with each other |
| 17:25.42 | [TK]D-Fender | [12:25]centurysami did use [jsmith1] but then extensions start going completely screwy in extensions.conf <- [jsmit1]'s calls go to the context definied in that device definition |
| 17:26.23 | centurysam | okay lets says there's two users. 100 and 101. if i setup [100] instead of jsmith1 i can call 102.. but if i use user1 / user 2 the extensions start messing up |
| 17:26.39 | centurysam | voicemail.conf also doesnt 'fetch username' but works when using ext. |
| 17:27.02 | WIMPy | Mailbox names are names that don't have to be numeric. |
| 17:27.20 | [TK]D-Fender | voicemail.conf ONLY supports numeric entries, not names. This is somewhat unfortunate for the sake of consistent naming possibilities |
| 17:27.29 | WIMPy | If you have trouble using non-numeric extensions of mailboxes, you're doing somethign wrong somewhere. |
| 17:27.47 | [TK]D-Fender | WIMPy: last I checked VM boxes are numeric-only |
| 17:27.50 | centurysam | understood. but again if i use mbox=jsmith@VoiceMail it doesnt grab username and fks up vmail when checking msgs. but if i set with mbox101@VoiceMail it works |
| 17:28.05 | [TK]D-Fender | centurysam: There is no username in there |
| 17:28.09 | [TK]D-Fender | centurysam: that is a BOX name |
| 17:28.18 | centurysam | yes well i set box name to username |
| 17:28.19 | WIMPy | VM names can be any string but they are limited in length. IIRC only 10 chars. |
| 17:28.29 | [TK]D-Fender | WIMPy: Certainly news to me. |
| 17:28.42 | [TK]D-Fender | WIMPy: I remeber consistent failures from others on that front before... |
| 17:29.05 | WIMPy | I'm using non-numeric VM names without any issues. |
| 17:29.06 | [TK]D-Fender | centurysam: Go show use your configs and the actual call failures. |
| 17:29.10 | [TK]D-Fender | ~pb |
| 17:29.10 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:29.11 | [TK]D-Fender | ^^^ |
| 17:29.29 | [TK]D-Fender | WIMPy: Well I guess that settles it. |
| 17:29.48 | *** join/#asterisk tenspeed705 (~keiths@goatfarm.vianet.ca) |
| 17:30.19 | centurysam | im looking for a basic config i can copy and know it works rather then try to config the crap out of my mess lol |
| 17:30.44 | centurysam | so rather then 101/101 it's jsmith/somepassword and has exten 101 and vmail works when checked |
| 17:30.56 | centurysam | right now i've got it all to the point where vmail just craps the bed |
| 17:32.19 | [TK]D-Fender | Show us |
| 17:33.50 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:36.24 | centurysam | http://pastebin.com/UmyQuqRa |
| 17:36.34 | centurysam | thats one users config |
| 17:37.02 | centurysam | i copy that to make another user and i can get them to connect but vmail when checked goes dumb to ComediaMail or whatever rather then password please |
| 17:37.44 | *** join/#asterisk jiuweigui (~jiuweigui@unaffiliated/jiuweigui) |
| 17:38.06 | centurysam | yes i could be doing it completely wrong, and would love some advice :) |
| 17:38.34 | [TK]D-Fender | Show us the call |
| 17:38.42 | centurysam | once second |
| 17:38.50 | WIMPy | Did you reload everything? |
| 17:39.26 | [TK]D-Fender | centurysam: And nothing there will cause it to ask for a password |
| 17:39.48 | [TK]D-Fender | centurysam: VOICMAILMAIN askes for a password (unless told otherwise), because that is the app you use to CHECK VM's. |
| 17:40.18 | [TK]D-Fender | centurysam: You showed us dialplan that happens to LEAVE voicemail whic is completely different and has no reason to ask. |
| 17:42.12 | centurysam | ok but i use the same setup on homebox and it works using 100 rather then shimmy |
| 17:42.12 | *** join/#asterisk Akuma (~Akuma@modemcable221.82-177-173.mc.videotron.ca) |
| 17:42.13 | centurysam | oh and here's msg when i dial |
| 17:42.13 | centurysam | v 10 12:40:25] WARNING[17985][C-0000000d]: app_voicemail.c:10545 vm_authenticate: Couldn't read username |
| 17:42.13 | centurysam | so it defaults to vmail rather then shimmy@VoiceMail |
| 17:42.13 | centurysam | ie user shimmy password doesnt work |
| 17:42.14 | [TK]D-Fender | I said "show us the call", not "show us some random singular line from CLI that I think is pertinent" |
| 17:42.17 | [TK]D-Fender | Showing us an error message doesn't let us see the command that CAUSED it. |
| 17:42.23 | centurysam | ok one second |
| 17:49.12 | centurysam | easiest way to capture one call? i dont want to pastebin 4 pages of sip debug |
| 17:51.44 | [TK]D-Fender | We don't mind |
| 17:52.43 | [TK]D-Fender | just do it |
| 17:52.43 | [TK]D-Fender | And there is no real way |
| 17:53.14 | WIMPy | And no need for sip debug. |
| 17:53.37 | centurysam | ok |
| 17:53.41 | centurysam | so it fixed itself |
| 17:53.46 | centurysam | no suddenly the password works |
| 17:54.02 | centurysam | fkn asterisk, does this shit all day.. same settings but only works after 5th reload |
| 17:54.31 | centurysam | !clear |
| 17:54.38 | WIMPy | Is you system time correct? |
| 17:54.52 | centurysam | yes |
| 17:54.59 | centurysam | it's done this on two seperate machines |
| 17:55.32 | centurysam | i copied a conf from one over and all of a sudden it didnt work, rewrote it the same way and it worked.. this really starts pissing you off when it happens constantly. |
| 17:55.44 | WIMPy | Asterisk does check the mtimes of the config files to avoid reloading ones that haven't changed. |
| 17:55.46 | centurysam | why does it suddenly work? eventhough it says unable to gather username? |
| 17:57.00 | centurysam | There any guide to a basic setup without having to use 'make samples' ? |
| 17:57.38 | WIMPy | ~primer |
| 17:57.38 | infobot | New to asterisk configuration? Check out this primer to get started. http://burner.com/asterisk-primer |
| 17:57.43 | WIMPy | ~book |
| 17:57.43 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:57.52 | centurysam | i can't ethnks |
| 17:57.53 | WIMPy | Try those |
| 17:58.01 | centurysam | grr. |
| 17:58.02 | centurysam | thnks |
| 17:59.27 | centurysam | wow, i knew coming here would fix my week long adventure of youtube and google. thnks again. |
| 18:00.42 | tenspeed705 | Anyone know what I can do to prevent a SIP device with forwarding enabled to not accept page requests? |
| 18:00.57 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 18:01.02 | WIMPy | Its manual. |
| 18:02.19 | tenspeed705 | WIMPy: was that directed to me? |
| 18:02.29 | WIMPy | yes |
| 18:02.46 | tenspeed705 | I don't think I understand what you mean? |
| 18:03.14 | [TK]D-Fender | [12:55]centurysamwhy does it suddenly work? eventhough it says unable to gather username? <- you have not shown us the call. You keep askign why and not showing the evidence. |
| 18:03.26 | WIMPy | Every device is different. You need to find out how the decive you're using dies it. If it does. |
| 18:03.29 | [TK]D-Fender | centurysam: If you want an autopsy, give us a body |
| 18:04.35 | [TK]D-Fender | [13:00]tenspeed705Anyone know what I can do to prevent a SIP device with forwarding enabled to not accept page requests? <- don't let users use taht feature on the phone iself. |
| 18:04.35 | tenspeed705 | Oh Ok. I thought It would just be something like using a SIP header. |
| 18:04.39 | [TK]D-Fender | tenspeed705: Or see if the phone offers a way to tell * about that state in a way it can do something with |
| 18:04.42 | [TK]D-Fender | tenspeed705: in SIP, "phone is king" |
| 18:04.52 | [TK]D-Fender | tenspeed705: Phone deos what then phone wants to do. |
| 18:05.07 | [TK]D-Fender | tenspeed705: And I've never ever seen an override like this before. |
| 18:05.13 | tenspeed705 | haha if it helps, they are Cisco SPA50X series phones |
| 18:05.24 | [TK]D-Fender | Go read their admin guide. |
| 18:05.27 | [TK]D-Fender | It'll tell you |
| 18:05.34 | tenspeed705 | downloading it now |
| 18:06.47 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 18:07.44 | *** join/#asterisk madax (~joshua@c187-193.i06-15.onvol.net) |
| 18:12.46 | [TK]D-Fender | tenspeed705: I seriously doubt they support this concept, but if you want to be sure... VERY FINE MANUALS |
| 18:16.42 | tenspeed705 | Found a better way, Looks like we use a macro for page, uses tge dial cmd. I have just added the i option to prevent forwarding |
| 18:17.19 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:19.28 | [TK]D-Fender | Wonder how the phone will react to that... |
| 18:21.33 | tenspeed705 | seems to be working fine. Talked to my senior admin, apparently thats why he built paging this way. See how it works going forward anyway |
| 18:41.39 | *** join/#asterisk CustosLimen (~CustosLim@unaffiliated/cust0slim3n) |
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| 19:36.59 | *** part/#asterisk KirilValchev (~KirilValc@ns.atsoftconsult-bg.com) |
| 19:38.24 | *** join/#asterisk areski (~areski@80.174.255.125.dyn.user.ono.com) |
| 19:40.09 | centurysam | interesting new issue: username is dialed rather then number when callback is pressed via remote phone rather then number.. should i just create 'name' extensions in dialplan? |
| 19:40.21 | centurysam | or can i define ext under username in sip.conf? |
| 19:41.08 | centurysam | i know i must be setting things up wrong lol |
| 19:48.37 | [TK]D-Fender | What your phone does when you press a button on it.. is up to your phoe |
| 19:48.39 | [TK]D-Fender | phone* |
| 19:50.26 | centurysam | no see what's messed up if i set [100] again in sip conf i call and they can select last call and call back without hitting 100 if i put jsmith then i have to define jsmith in dialplan for them to be able to call back |
| 19:50.48 | centurysam | without dialing 100 but rather pressing on call history callback |
| 19:51.39 | [TK]D-Fender | Sounds like you didn't define a CALLER-IR on your devices so that it has a NUMBER to associate to that incoming call. |
| 19:51.48 | [TK]D-Fender | You are missing having done the basics.... |
| 19:52.07 | centurysam | ah, okay so now callerid is needed when using jsmith |
| 19:52.11 | centurysam | but not when just using 100 |
| 19:52.14 | WIMPy | That description was too vague for me to say anything. |
| 19:52.18 | centurysam | lol see why this frustrates me? |
| 19:52.20 | file | falls over |
| 19:52.31 | WIMPy | No. |
| 19:52.35 | centurysam | yes |
| 19:52.42 | centurysam | hold on on ill explain |
| 19:52.53 | WIMPy | It's needed so the caller knows who's calling. |
| 19:53.02 | [TK]D-Fender | SET THE CALLERID ON YOUR SIP.CONF ENTRIES FOR YOUR PHONES. |
| 19:53.04 | [TK]D-Fender | ^^^^^^^^^^^ |
| 19:53.10 | WIMPy | It still doesn't matter what your extensions or usernames look like. |
| 19:53.12 | [TK]D-Fender | how is that on the clarity scale? |
| 19:53.20 | [TK]D-Fender | Forget username |
| 19:53.29 | [TK]D-Fender | Set the CALLERID on them |
| 19:53.31 | centurysam | lol |
| 19:53.33 | centurysam | ok |
| 19:53.42 | centurysam | i know what you mean but im explaining how it's been acting |
| 19:53.46 | [TK]D-Fender | Username is NOT some super-centric magical value for everything |
| 19:53.48 | WIMPy | And if you don't set calerid in your sip.conf, Asterisk will forward what you configured on the calling phone. |
| 19:53.53 | centurysam | maybe it works well in a perfect world but it seems ive killed that world |
| 19:54.12 | centurysam | i get what your saying |
| 19:54.16 | [TK]D-Fender | It works in the world where the user actualy sets up the normal stuff everyone should be doing. |
| 19:54.22 | centurysam | but without callerid and in sip its just 100 it works |
| 19:54.31 | centurysam | if its jsmith suddenly you need callerid |
| 19:54.33 | centurysam | or it doesnt |
| 19:54.36 | centurysam | atleast thats what its doing |
| 19:54.37 | [TK]D-Fender | becasue thta is the ONLY thing it sees about the incoming call |
| 19:54.40 | [TK]D-Fender | and it HAPPENSQ to match |
| 19:55.06 | [TK]D-Fender | So stop taking this "luck" as being "standard operating procedure" |
| 19:55.09 | centurysam | so it should be smart enough to realize to call my exten rather then my username |
| 19:55.10 | WIMPy | Did you change your phones config as well? If no, then that' what's going wrong. |
| 19:55.19 | centurysam | yes |
| 19:55.22 | WIMPy | You want to avoid havong to do so? Set it in Asterisk instead. |
| 19:55.24 | [TK]D-Fender | He shouldn't be letting his devices dictate this anyway |
| 19:55.28 | centurysam | im using jmith1 rather then 100 |
| 19:55.31 | centurysam | i can call |
| 19:55.33 | centurysam | get calls |
| 19:55.41 | [TK]D-Fender | Take responsibility and configure your server properly |
| 19:55.45 | centurysam | but click log and callback it dials username jsmith1 not 100 |
| 19:55.48 | centurysam | anyway |
| 19:55.51 | centurysam | ill add callerid |
| 19:55.58 | centurysam | rather then argue lol :) |
| 19:56.16 | [TK]D-Fender | that SAYS what to make calls from that device look like |
| 19:56.20 | doop | let's all browbeat each other into best practices together |
| 19:57.10 | WIMPy | Shopuldn't we start with best theories first, before practicing? |
| 19:58.13 | centurysam | no just stone me for asking questions lol |
| 19:59.10 | WIMPy | In IRC it's more common to use large trouts. |
| 20:00.09 | doop | only if you run windows |
| 20:00.43 | WIMPy | That's probably true. |
| 20:08.08 | centurysam | so okay, if no callerid is set it pushes jsmith1 but if i put number in callerid it no longer pushes jsmith1 but the extension im calling from.. whew |
| 20:08.34 | WIMPy | No. Whatever you wrote there. |
| 20:09.09 | centurysam | without callerid under jsmith1 it does push jsmith1 as callerid. if i use callerid under jsmith1 it'll push number |
| 20:09.14 | centurysam | that's exactly what happens |
| 20:09.48 | WIMPy | Not really. I already tried to tell you. |
| 20:10.03 | centurysam | yes really, because thats just what happened |
| 20:10.11 | centurysam | rather then jsmith1 showing up a number 100 showed up |
| 20:10.14 | centurysam | and i could callback |
| 20:10.15 | centurysam | lol |
| 20:10.19 | WIMPy | That's what seems to have happened. |
| 20:11.26 | centurysam | this shit is more tempermental then my ex! |
| 20:11.50 | WIMPy | Wait until you get to the interesting bits. |
| 20:12.09 | centurysam | lol im already scared and i only have two ppl defined |
| 20:13.35 | *** join/#asterisk areski (~areski@80.174.255.125.dyn.user.ono.com) |
| 20:15.19 | centurysam | im still stuck with how i dial "1" to check voicemail, rather then say password please. it shoots Comedian Mail.... Mailbox...... password please |
| 20:15.31 | centurysam | so i have to wait out until it tries to id me |
| 20:15.41 | centurysam | all because of jsmith1 rather then 100 |
| 20:18.01 | centurysam | Anybody got a clue on why it spits out Comedian Mail... mailbox.. rather then saying password without intro |
| 20:18.26 | doop | are you passing the mailbox number to the voicemail app as an argument when calling it in the dialplan? |
| 20:18.31 | centurysam | i assume i must change voicemail.conf to use ext rather then username? |
| 20:18.38 | WIMPy | You didn't tell it what mailbox you want. |
| 20:19.08 | centurysam | damnit, all i do is change to jsmith rather then 100 and it efs up |
| 20:19.12 | centurysam | so whats going on |
| 20:19.24 | centurysam | i should use ext in mbox and voicemail? |
| 20:19.49 | centurysam | callerid set |
| 20:19.53 | centurysam | lol cant be that now |
| 20:20.01 | centurysam | it should know its me |
| 20:20.13 | centurysam | and infact doesnt give username error like before |
| 20:20.20 | centurysam | due to callerid being set |
| 20:20.24 | WIMPy | didn't know it could find out on its own. |
| 20:20.40 | centurysam | yep solved it |
| 20:20.45 | doop | in fact, this electronic "brain" practically thinks for itself !!!!1 |
| 20:20.53 | centurysam | again you cannot use names in vmailbox |
| 20:21.19 | SQUelcher | bye |
| 20:21.20 | centurysam | doop: it's retarded then |
| 20:21.35 | doop | centurysam surely you're aware of sarcasm |
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| 20:30.05 | centurysam | damn, vmail works now too |
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| 23:19.57 | mrbnet | I have an extension setup which rings multiple SIP endpoints. Is there a function I can call in an AGI script to identify which one answered the call? |
| 23:21.43 | WIMPy | 'core show application dial' |
| 23:22.01 | WIMPy | DIEALEDPEERNAME or something. |
| 23:23.14 | przerull | you might try getting that from the agi_dnid or agi_channel that is set when the agi session is set up |
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| 23:27.31 | mrbnet | The issue with all of the variables I have tried is that they only show me the extension which was dialed. That extension dials (Local\101&Local\102&SIP\4352) How can I tell which one of those picked up the call? |
| 23:28.55 | WIMPy | Your wording is bad. extension or list of extensions? |
| 23:29.07 | WIMPy | (put aside that they are not extensions in Asterisk terms) |
| 23:30.29 | WIMPy | DIALEDPEERNAME and DIALEDPEERNUMBER are the ones telling you who answered. |
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| 23:34.02 | mrbnet | WIMPy: Thanks! Missed the dial channel variables in the docs. Those are just what I needed. |
| 23:40.02 | przerull | so I'm configuring asterisk 13.0.0 for webrtc. my user agent (jssip) is able to register fine but I keep getting 488's for my invites. I've followed the instructions on https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support and did a fair bit of googling but i'm stuck |
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| 23:43.10 | przerull | here's my pjsip.conf https://gist.github.com/anonymous/7d52185220ef6c7f5993 |
| 23:46.52 | przerull | according to https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support it's suggested that either avpf support is not enabled or have not allowed a codec that is supported by the caller but from pjip.conf we are enabling avpf and ulaw |