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| 00:46.00 | DBordello | I am trying to install res_xmpp. make menuconfig shows: Depends on: iksemel(E) |
| 00:46.00 | DBordello | <PROTECTED> |
| 00:46.21 | DBordello | So i installed libiksemel-dev and libssl-dev. However, after a ./configure, make menuconfig shows the same. Ideas? |
| 00:51.12 | robert_ | !pastebin |
| 00:52.37 | WIMPy | ~pb |
| 00:52.37 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 00:58.24 | jeev | is it ok to ask about certain ITSP's here? |
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| 01:01.52 | igcewieling | ask. if you don't get kickbanned, you'll know the answer |
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| 01:06.34 | robert_ | WIMPy: thanks. :D |
| 01:09.05 | DBordello | Any hints on getting res_xmpp to compile |
| 01:10.43 | jeev | :> |
| 01:10.45 | jeev | anyone heard of geils? |
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| 01:20.10 | robert_ | http://dpaste.com/1205664/ -- so this is our sip subscribers table. I'm trying to figure out a couple things there. One, for some reason, it doesn't think 2000 or 3000 are available when I 'sip show [peer-id] load', two, I don't know which context settings puts a SIP subscriber inside a specific context so I get insider-only extensions and such. This is the context I'm talking about, http://dpaste.com/hold/1205669/; any ideas on why it isn't "seeing" |
| 01:20.11 | robert_ | <PROTECTED> |
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| 01:26.19 | DBordello | Is it possible just to make the modules, and not the whole burrito? I am on a slow processor |
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| 01:30.11 | WIMPy | The Burrito is made of modules. |
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| 01:30.22 | DBordello | Then the whole burrito it is then |
| 01:30.45 | DBordello | i assumed they were seperate files |
| 01:31.04 | WIMPy | they are |
| 01:31.08 | WIMPy | But they need to fit. |
| 01:31.22 | DBordello | okay. The poor raspberry pi is chewing on them now |
| 01:31.23 | WIMPy | Apart from that the usual makefile magic applies. |
| 01:32.20 | DBordello | fair enough |
| 01:32.26 | DBordello | make distclean sent me back to the start |
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| 01:33.06 | WIMPy | That's what it does. |
| 01:33.22 | DBordello | I was desperate. Looking back, I don't think it was necessary |
| 01:34.03 | WIMPy | I've never done it. |
| 01:34.42 | WIMPy | I think I needed a 'make clean' once, but that was quite some time ago. |
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| 01:53.10 | DBordello | waits and waits |
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| 01:53.37 | pickleheadjones | Anyone alive? |
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| 02:12.43 | pickleheadjones | Anyone using Asrterisk in a cloud environment? |
| 02:13.14 | pickleheadjones | Anyone using Asrterisk in a cloud environment? |
| 02:13.42 | pickleheadjones | Ok Cool |
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| 02:38.46 | DBordello | pickleheadjones, ask your question |
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| 03:03.11 | DBordello | robl^, Where did you see the book for version 11? |
| 03:03.30 | DBordello | Nevermind, ignore. |
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| 03:23.34 | DBordello | If I want to send a DTMF "1" to the calling party, I am trying this dialplan: same => n,Dial(SIP/djb-desktop,20,D(,1)) |
| 03:23.49 | DBordello | However, that sends ",1" to the called party. What is the proper syntax? |
| 03:27.33 | DBordello | Ah, it is a colon to seperate parameters |
| 03:28.17 | robert_ | anybody? lol |
| 03:28.42 | DBordello | robert_, questioning how I am talking to myself? :) |
| 03:28.48 | robert_ | :p |
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| 03:30.10 | robert_ | nah |
| 03:30.14 | robert_ | I have a question above :p |
| 03:30.37 | DBordello | I am in no position to be offering advice |
| 04:09.18 | ChannelZ | I didn't see your question |
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| 04:11.27 | ChannelZ | didn't/don't |
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| 04:29.21 | DBordello | <robert_> http://dpaste.com/1205664/ -- so this is our sip subscribers table. I'm trying to figure out a couple things there. One, for some reason, it doesn't think 2000 or 3000 are available when I 'sip show [peer-id] load', two, I don't know which context settings puts a SIP subscriber inside a specific context so I get insider-only extensions and such. This is the context I'm talking about, http://dpaste.com/hold/1205669/; any ideas on w |
| 04:29.22 | DBordello | hy it isn't "seeing" |
| 04:29.22 | DBordello | <robert_> [employee]? I know I need to migrate over to realtime, however the issue is that Ast keeps putting SIP users inside the 'default' context and not the one specified. When making a call say, to *4357, it will reject due to the extension not being found in [default]. |
| 04:29.27 | forst | Does anyone here have experience unbricking Mitel 5xxx phones? |
| 04:30.02 | forst | Had a firmware update kill about 50 phones last night, the ethernet port is pretty much useless even after defaulting. "bad lan link" |
| 04:30.18 | forst | so I'm trying to figure out a way to maybe reflash the chips manually |
| 04:33.07 | DBordello | Ouch |
| 04:35.35 | forst | yeah it sucked, had to replace all of the dead phones with brand new phones to get the customers back up and running.. ~$9000 disaster lol |
| 04:36.38 | forst | I guess this isn't the best place to ask about reflashing them though, I just don't know which channels would discuss stuff like that, jtag etc.. any ideas? |
| 04:36.49 | DBordello | I do not, sorry :/ |
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| 04:39.59 | ChannelZ | Well it suggests that the device making the call is not actually matching the peer you think it is |
| 04:40.26 | ChannelZ | But I'm guessing without any evidence |
| 04:41.11 | apb1963 | forst: http://www.mitel.com/partners/mitel-user-group/ |
| 04:41.24 | ChannelZ | (the 'context' column is the context the device should be in) |
| 04:43.01 | ChannelZ | robert_: oh.. actually just looking at your INSERT statements you're not even setting context on 2000,2001,3000 so that explains that |
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| 04:56.32 | ChannelZ | So, in Google's transition of Voice/Talk into Hangouts, have they changed or otherwise busted the protocol? |
| 05:18.53 | kaldemar | ChannelZ: hangouts does not support XMPP. |
| 05:20.35 | ChannelZ | Yeah I'm just finding some writeups on that. So does Motif becomes useless at some point down the road? Going the way of Skype? |
| 05:26.13 | kaldemar | they didn't explicitly say anything about that but since XMPP services are replaced by something else, you can do the math... |
| 05:26.36 | ChannelZ | ...Google... feh |
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| 05:40.37 | reenignEesreveR | whats the best way to get a REST API on top of asterisk? |
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| 06:56.06 | din3sh | hello all |
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| 07:04.21 | bulkorok | hi |
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| 07:07.45 | din3sh | hello, anyone been able to set up billing using Channel Event Logging? CEL? |
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| 07:26.38 | jnemeth | Is there anybody that can help with an issue with outdialing on a PRI? I can receive calls on the PRI just fine. |
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| 07:32.33 | din3sh | jnemeth: what is the problem? |
| 07:32.49 | din3sh | i mean what error message you having? |
| 07:33.45 | jnemeth | in response to Dial, it gives this error: [May 27 02:40:06] WARNING[3923] app_dial.c: Unable to create channel of type 'DAHDI' (cause 6 - Channel unacceptable) |
| 07:34.27 | jnemeth | I've been busy with Google, but not having much luck figuring out what the error means. |
| 07:35.08 | jnemeth | Since I can receive calls, the configuration must be mostly correct. |
| 07:37.34 | jnemeth | the actual call to Dial was Dial(DAHDI/g1/${EXTEN}) |
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| 07:38.21 | din3sh | and? |
| 07:38.32 | din3sh | error msg? |
| 07:38.48 | din3sh | what does log say? |
| 07:39.19 | jnemeth | see above... |
| 07:39.53 | jnemeth | immediately after the line where you first asked for the error message... |
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| 08:26.54 | WIMPy | jnemeth: Looks like that group isn;t set up correctly. |
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| 08:29.42 | jnemeth | WIMPy: having just looked at chan_dahdi.conf.sample (which is ridiculously long) again, I was just wondering about that... it currently has: |
| 08:29.43 | jnemeth | channel=>1-23 |
| 08:29.43 | jnemeth | group=1 |
| 08:30.01 | jnemeth | I'm now thinking that those two should be reversed... |
| 08:30.19 | kaldemar | they should, if you want channels 1-23 to belong to group 1. |
| 08:31.11 | kaldemar | in chan_dahdi.conf, a setting applies for all channels defined below it, until the setting is defined otherwise. |
| 08:31.56 | jnemeth | yes, I'm going to move the channel line to the bottom and try again... |
| 08:32.37 | jnemeth | I can't try right now as I'm not on site and don't have easy access... |
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| 08:33.06 | WIMPy | Yes, the chan_dahdi.conf is some very bad spaghetti. |
| 08:40.20 | din3sh | Hey WIMPy |
| 08:42.04 | WIMPy | waves |
| 08:46.19 | din3sh | how is it going? |
| 08:46.20 | din3sh | :D |
| 08:47.49 | WIMPy | A bit like chan_dahdi.conf. Not very well organized. |
| 08:48.01 | din3sh | hahaha |
| 08:48.06 | jnemeth | hahah |
| 08:48.21 | WIMPy | With a certain loss of overview. |
| 08:48.28 | din3sh | like the CDR logging is well organised |
| 08:48.59 | din3sh | :] |
| 08:49.16 | WIMPy | Like missing the the part that's usually the reason for having them? |
| 08:49.28 | din3sh | yeah |
| 08:49.41 | din3sh | trying to set up billing with the CDR |
| 08:50.01 | WIMPy | Yes, pretty bad :-( |
| 08:50.19 | din3sh | wont be able to do that, CDR is erroneous for atxfer etc |
| 08:50.47 | WIMPy | I remember when I first used Asterisk and wrote some scripts to combine Asterisks CDRs wit the CDRs of the PBX behind. |
| 08:51.23 | din3sh | in 1.4 i found some agi script to do that |
| 08:51.34 | din3sh | not in 1.8 or 11 |
| 08:52.11 | WIMPy | Trouble is that the reason for writing CDRs is missing: The costs. |
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| 08:53.26 | din3sh | Phone A calls Phone B, transfers to Phone C. There is no CDR log for B talking to C, even if its talking for 2hrs |
| 08:53.28 | din3sh | :o |
| 08:54.13 | din3sh | has been a major issue since 2008. I dont think there's a fix for that even as of today |
| 08:54.19 | jnemeth | that looks like it's going to be fun... |
| 08:54.21 | WIMPy | Luckily at the point where I needed them transfers would not happen in Asterisk. |
| 08:54.43 | msaraiva | din3sh: channel event logging. |
| 08:54.53 | msaraiva | Solves all your problems like magic. |
| 08:54.59 | din3sh | tried that |
| 08:55.07 | jnemeth | Asterisk: The Definitive Guide seems to recommend CEL logging to solve that problem... but, those logs look like they will be a lot harder to analyze... |
| 08:55.17 | din3sh | i now have 10 records in the db for a single extension to extension call |
| 08:55.32 | din3sh | FOR A SINGLE CALL |
| 08:55.54 | din3sh | for transfers you have like 20 records for a 1 legged transfer |
| 08:55.58 | din3sh | o.O |
| 08:56.25 | WIMPy | Yes, I guess the first setup (many years ago) is still the best. Use your old PBX and just add Asterisk for some extra features. |
| 08:57.00 | jnemeth | doesn't work so well when there isn't an "old PBX"... |
| 08:57.18 | din3sh | https://issues.asterisk.org/jira/browse/ASTERISK-11309 |
| 08:57.19 | LieutPants | [ASTERISK-11309] [Status: Closed] Missing CDR's for Transfers - https://issues.asterisk.org/jira/browse/ASTERISK-11309 |
| 08:57.30 | WIMPy | jnemeth: ebay :-) |
| 08:58.11 | jnemeth | yeah, that just complicates things greatly... |
| 08:58.40 | WIMPy | The sensible possiblities for connecting local phones are rater limited anyway :-( |
| 09:00.48 | jnemeth | hrmm: Resolution: Won't Fix |
| 09:01.15 | jnemeth | new installs equal SIP phones... |
| 09:01.57 | WIMPy | SIP phones are ony an option if you are in a place with (almost) free energy. |
| 09:02.03 | WIMPy | only |
| 09:04.38 | WIMPy | Will be a good business to tell people how much money they can save when they change to a classic PBX. |
| 09:05.38 | din3sh | classic PBX? |
| 09:05.41 | din3sh | like avaya? |
| 09:05.43 | din3sh | :p |
| 09:06.02 | WIMPy | Whatever. Something with non-IP phones. |
| 09:06.50 | din3sh | digital ones? |
| 09:07.30 | WIMPy | I thought analoge ones have been extinct for soe decades. |
| 09:07.32 | din3sh | are SIP phones a regression to these ones? |
| 09:07.33 | din3sh | :D |
| 09:07.38 | WIMPy | some |
| 09:07.45 | WIMPy | Yes |
| 09:09.04 | WIMPy | They are just like your mobile smartphones. You can do lots of nice things with them, but making calls isn;t as easy as it used to be. |
| 09:10.37 | din3sh | Ah! |
| 09:10.48 | din3sh | i have an android based sip phone on my desk |
| 09:11.00 | din3sh | a pain in the ass to do simple thing as dialing a number |
| 09:12.54 | din3sh | with a digital phone, when you call out an extension, you used to see the name+number you are calling |
| 09:13.14 | din3sh | with *, you have to set connectedID() |
| 09:13.25 | jnemeth | 6.9 cents per kWh for the first 1354 kWh, then 10.34 cents per kWh for the rest, where I am... |
| 09:13.54 | WIMPy | Names actually being transmitted instead of just being inserted via your local directory ar a real advantage. |
| 09:14.26 | *** join/#asterisk felimwhiteley_ (~quassel@89.101.203.26) |
| 09:14.42 | jnemeth | yeah, but * is infinitely flexible; all that can be fixed :-> |
| 09:14.57 | WIMPy | In the standard plan you're at 29.12 ยข/kwh here. |
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| 09:15.23 | WIMPy | jnemeth: Err, no. Already forgot the CDR thing? |
| 09:16.36 | jnemeth | expensive power you have over there... |
| 09:17.04 | WIMPy | Yes. That's why I think there's a business opportunity. |
| 09:17.59 | jnemeth | are you using POE with your SIP phones? or, why do your SIP phones use so much power? |
| 09:18.16 | WIMPy | With a SIP phone you easily use 100-200 times more electricity than with a digital PBX or ISDN phone. |
| 09:18.50 | WIMPy | I personally use only one SIP phone for that very reason. |
| 09:19.23 | jnemeth | seems rather strange... a SIP phone shouldn't be any more complex then an ISDN phone electronically... |
| 09:20.09 | WIMPy | Oh, Ethernet is very bad regarding power consumption. |
| 09:20.43 | jnemeth | once upon a time, I did a work term in a lab that was designing ISDN phones, so I have a pretty good idea what goes into them... |
| 09:21.39 | jnemeth | hrmm... |
| 09:22.04 | WIMPy | And the above figure incluses porportional amounts of power for switches, off course. |
| 09:23.16 | jnemeth | yeah, but the * box itself should use a lot less power then a classic PBX... |
| 09:23.35 | jnemeth | would quite possibly use more power then a small key system... |
| 09:23.42 | WIMPy | If you run it on a Raspberry PI. |
| 09:24.20 | jnemeth | raspi don't have much oomph to them... that would be a rather low capacity system... |
| 09:24.35 | WIMPy | Indeed. |
| 09:24.56 | jnemeth | when you say PBX, do you mean an actual PBX or a key system (what you find in most small offices)? |
| 09:25.36 | WIMPy | PBX. Key systems don't really exist here. Certainly not since the 80s. |
| 09:25.55 | jnemeth | an actual PBX would like use a lot more power then most * systems... |
| 09:26.09 | WIMPy | No |
| 09:28.06 | WIMPy | My old soho PBX used 11W including 5 phones. |
| 09:28.28 | WIMPy | And the bigger ones aren't as bad as you might think, either. |
| 09:31.01 | jnemeth | around here, Meridian Norstar systems are quite popular... they are fully electronic, but they are generally considered to be key systems... |
| 09:32.03 | *** join/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 09:34.27 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
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| 09:36.43 | jnemeth | most small office systems are what I and most people I know would classify as key systems (basically they have seperate line buttons for each CO line)... |
| 09:37.17 | jnemeth | those type of systems are still very common... |
| 09:38.09 | jnemeth | the advent of * and other lower cost voip systems is slowly changing that though... |
| 09:39.35 | WIMPy | You actually find offices here that just use cordless phones as DECT bases always come with mini PBX included. |
| 09:39.49 | WIMPy | Not my cup of tea, but seems to work. |
| 09:40.59 | WIMPy | That's the version where the base is incluses in the only desktop phone, usually. |
| 09:42.27 | linocisco | if one PSTN line is down, how can users make automatically select next working lines? |
| 09:43.13 | WIMPy | Use groups or DIALSTATUS. Depends on what kind of lines. |
| 09:43.25 | WIMPy | And what kind of "down". |
| 09:48.12 | din3sh | linocisco: man you still with that :p |
| 09:48.25 | din3sh | switchover to UCM |
| 09:48.27 | din3sh | :D |
| 09:49.00 | linocisco | WIMPy, no. it is down by PSTN companies |
| 09:49.02 | kaldemar | linocisco: you better start using a technology specific term instead of "PSTN line". |
| 09:49.33 | linocisco | kaldemar, what technology specific term? we dont have E1 or T1 or ISDN or good internet |
| 09:50.11 | kaldemar | linocisco: PSTN is not a specific technology. if you refer to analog, use FXO or FXS. |
| 09:50.32 | linocisco | kaldemar, ok. analog FXO land lines |
| 09:50.33 | kaldemar | maybe you're mixing PSTN and POTS. |
| 09:51.12 | din3sh | DIALSTATUS is the way to go linocisco, as i told u 2 weeks back |
| 09:51.14 | din3sh | :] |
| 09:51.41 | linocisco | kaldemar, what is the different between? |
| 09:52.09 | WIMPy | ~pstn |
| 09:52.09 | infobot | it has been said that pstn is Public Switched Telephone Network, or "please stop the nonsense" |
| 09:52.11 | WIMPy | ~pots |
| 09:52.11 | infobot | [~pots] POTS (Plain Old Telephone Service) is the term for a common analog phone line service as is used world-wide. The "phone company" is called FXO (~fxo), and the user end-point (or phone) is called FXS (~FXS). POTS supports 1 channel, and possibly call-waiting, 3-way calling, CID, as signalled to the telco. |
| 09:53.13 | WIMPy | wonders what's nonsense about the PSTN. Apart from it obviousely not being maintained any more in some areas :-( |
| 09:53.50 | din3sh | "please stop the nonsense" seems to be a better definition |
| 09:54.53 | linocisco | WIMPy, we dont have 3 way calling |
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| 09:55.11 | linocisco | WIMPy, no call waiting |
| 09:55.58 | WIMPy | Bad, but doesn;t matter as you couldn't make much use of that anyway when using Asterisk. |
| 09:56.09 | kaldemar | linocisco: Plain Old Telephone Service (=analog technology) and Public Switched Telephone Network (=network). latter is a network that uses different technologies, POTS just refers to analog technology (=FXS/FXO). |
| 09:57.19 | linocisco | WIMPy, but i can use full features of asterisk like call wating, call forward, transfer, meetme, voicemail, directory, for all extensions |
| 09:57.58 | WIMPy | But none of them on the network side. |
| 09:59.07 | jnemeth | anyways, thanks for the help earlier... but, it is 3 AM local time and long past time I went to bed... |
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| 10:26.38 | linocisco | WIMPy, all internally |
| 10:27.54 | *** join/#asterisk Gr3mlin (7deeffc4@gateway/web/freenode/ip.125.238.255.196) |
| 10:27.59 | Gr3mlin | hay guys |
| 10:28.06 | Gr3mlin | hows everyones evening going? |
| 10:28.29 | WIMPy | Ask later :-) |
| 10:28.55 | Gr3mlin | passes WIMPy his last Jim Beam |
| 10:29.53 | WIMPy | I hope you added lots of cola to make it drinkable. |
| 10:30.48 | Gr3mlin | i did indeed! really cola too, not RC-cola |
| 10:34.52 | *** join/#asterisk War_Bear (~War_Bear@warbear.co.uk) |
| 10:35.19 | Gr3mlin | i did have a Asterisk related question, hopefully a quick one, Q: "I've setup a Virtual Server running Asterisk, and Asterisk Gui, to play with, before committing a PC too it. only, when i access the Gui, i get the lovely "Checking write permission for gui folder" error. now i've googled the poo out of it. but still i am stuck on it. Is this simply a Hardware issue as the VM doesnt have a PSTN device of any kind?!" |
| 10:36.22 | WIMPy | No |
| 10:36.48 | WIMPy | But You will have a hard time finding support for the Asterisk GUI. |
| 10:37.40 | Gr3mlin | i thought it might be easier to use the gui than trying to wrap my head around it. |
| 10:38.34 | WIMPy | IF you really want a GUI, try FreePBX. It's not supported here, either, but at least there is some support in #freepbx. |
| 10:38.48 | Gr3mlin | yeah? is that cos its this is #asterisk and not #asterisk-gui? or just the gui is overrated!? |
| 10:39.32 | WIMPy | However you might consider that any GUI will limit yur possibilities and is not a good starting point if you intend to learn Asterisk yourself. |
| 10:42.17 | Gr3mlin | to be honest, i though i would have a little play, as i need to make a landline accessible via an IP phone located in a different location. |
| 10:42.57 | Gr3mlin | so as far as i can tell i will be using 1% of asterisks potential. |
| 10:43.45 | WIMPy | The GUIs should be more than adequate for that task. |
| 10:44.08 | din3sh | use elastix or asterisknow man |
| 10:44.23 | din3sh | or freepbx (all of these are freepbx) |
| 10:44.30 | din3sh | less pain the ass |
| 10:44.32 | WIMPy | Elastix??? |
| 10:44.33 | din3sh | in the * |
| 10:44.46 | din3sh | yes |
| 10:45.01 | WIMPy | You mean elastix as in |
| 10:45.05 | WIMPy | ~elastix |
| 10:45.05 | infobot | somebody said elastix was a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
| 10:45.18 | din3sh | yeah |
| 10:45.26 | din3sh | have tried it |
| 10:45.59 | Gr3mlin | <-- is downloading FreePBX right now :D |
| 10:46.03 | WIMPy | I guess you have better chance to get support with AsteriskNOW and FreePBX? |
| 10:46.10 | din3sh | Elastix is pretty popular in spanish community, south american countries etc |
| 10:46.21 | din3sh | they even have their own turnkey appliance |
| 10:46.35 | din3sh | you have paid support with elastix also |
| 10:46.46 | din3sh | together with training/certification |
| 10:47.22 | WIMPy | Interesting. |
| 10:47.30 | Gr3mlin | Sounds like Concept Security panels. 1500$ just for a piece of paper saying to attended 2 weeks of training. |
| 10:47.50 | din3sh | but i havent followed any training/certification |
| 10:47.50 | din3sh | :D |
| 10:48.07 | din3sh | my past handson with pure asterisk helps me |
| 10:48.33 | din3sh | WIMPy i deployed elastix with their free call centre module with 70agents |
| 10:48.36 | din3sh | pretty much ok |
| 10:48.48 | Gr3mlin | i know people that self teach, and know more that 'Qualified' people. |
| 10:49.56 | Gr3mlin | OOoo can someone point me to the recommend hardware to connect to the PSTN line? |
| 10:50.00 | WIMPy | Isn't that true most of the time. |
| 10:50.12 | WIMPy | Gr3mlin: What kind of line? |
| 10:50.35 | Gr3mlin | standard telephone? |
| 10:50.48 | Gr3mlin | is that what you mean WIMPy ? |
| 10:51.01 | WIMPy | Buy a digium card or a SIP gateway. |
| 10:52.08 | Gr3mlin | why do they seem to always be rigged for 4 analog lines? |
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| 10:55.14 | Gr3mlin | what about a Authentic X100P WIMPy |
| 10:56.22 | WIMPy | Analog stuff is bad enough. Don't give yourself the pain of a cheap clone. |
| 10:58.55 | Gr3mlin | really? plopper. ok. might have to tell the boss to s-can it. trying to do it as cheap as i can for him. |
| 11:01.15 | din3sh | cheap? |
| 11:01.36 | din3sh | cheap sip phones will give u sleepless nights |
| 11:01.37 | din3sh | :D |
| 11:01.41 | WIMPy | Analog stuff doesn't come cheap. |
| 11:01.45 | Gr3mlin | ooo, what about this?? can i post links? |
| 11:02.10 | WIMPy | yes |
| 11:02.31 | Gr3mlin | http://www.trademe.co.nz/electronics-photography/phone-fax/other/auction-597845218.htm |
| 11:03.25 | WIMPy | Didn't I say buy Digium? |
| 11:04.14 | Gr3mlin | indeed, i just found that one :D |
| 11:05.29 | Gr3mlin | 75 or 65.. i think i will pay the extra 10 ;) |
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| 11:11.48 | Gr3mlin | thanks for the help guys! :) Have a good evening / day / weekend |
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| 12:04.32 | skrusty | afternoon |
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| 12:46.08 | zafu | hi, is there some secret ingredient to make a 7960 register itself through NAT (I did set the proper settings)? |
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| 13:07.38 | Free99 | hey, if I needed to reload the context of a single peer, how would I do that? |
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| 13:09.54 | [TK]D-Fender | Whole dialplan is loaded in one shot. |
| 13:10.01 | [TK]D-Fender | And has nothing to do with your peers |
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| 13:12.39 | Free99 | [TK]D-Fender, issue is that I have active clients but I had to modify dialplan. My DID has a specific context that it seems to forget to jump to unless I do a full reload |
| 13:13.00 | Free99 | the DID being statically defined in sip.conf |
| 13:13.17 | Free99 | or not DID, the termination provider |
| 13:13.30 | [TK]D-Fender | the SIP entry? |
| 13:13.41 | [TK]D-Fender | Do not mix PEERS with DIALPLAN |
| 13:13.52 | [TK]D-Fender | and "forget to jump" is not the case |
| 13:14.05 | [TK]D-Fender | And there is no "partial" |
| 13:14.48 | Free99 | So how should I do this then? I have a peer defined in sip.conf with a certain context=.... when I do dialplan reload, the peer stops working properly.. |
| 13:15.18 | Free99 | the default context is different than the one this peer needs to go to |
| 13:15.34 | Free99 | when I do a full reload, the peer works again |
| 13:16.28 | Free99 | [TK]D-Fender, so is that a bug then? |
| 13:16.40 | [TK]D-Fender | you are mistaken about the circumstances. |
| 13:17.01 | [TK]D-Fender | The peer points to wherever it points to. If you change the dialplan and reload your next will be with the new dialplan |
| 13:17.45 | [TK]D-Fender | next call* |
| 13:18.17 | Free99 | you're saying that the context to which the peer points should still work as it does normally, despite the dialplan reload? |
| 13:19.33 | [TK]D-Fender | ? |
| 13:19.45 | [TK]D-Fender | new call loads the CURRENT dialplan. |
| 13:19.49 | [TK]D-Fender | You are talking in circles |
| 13:19.54 | [TK]D-Fender | There is no magic to this |
| 13:22.42 | Free99 | [TK]D-Fender, just hear me out for a sec: What it seems to me you are saying is, if I set context=<existant context in extensions.conf> for the peer, then start asterisk... the peer will behave as expected right? It seems like you are also saying that if I change the dialplan in a different area, *not* in the context for that peer and do a dialplan reload, the peer should still behave as expected, correct? |
| 13:28.32 | Free99 | sigh. I wish I knew how to setup the jitter buffer :-/ |
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| 13:30.51 | Free99 | well ok [TK]D-Fender maybe I'm barking up the wrong tree. Let me ask this then: is there a way to do a reload without losing all my registrations? |
| 13:33.31 | jmetro | have your registrations done through a separate sip registration server |
| 13:34.17 | [TK]D-Fender | Sialplan has nothing to do with registrations |
| 13:34.33 | [TK]D-Fender | Neither does reloading SIP |
| 13:34.50 | [TK]D-Fender | If you're registered then that fact is held |
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| 13:42.07 | Free99 | [TK]D-Fender, I believe perhaps there's a bug in that case, because that is not what happens. |
| 13:42.31 | [TK]D-Fender | Free99: Doubt it highly |
| 13:43.08 | igcewieling | Free99: Yes. No. Without more information it is impossible to tell. for example, are you using Realtime |
| 13:43.34 | Free99 | igcewieling, yes I am. This peer however is statically defined in sip.conf |
| 13:43.57 | igcewieling | Free99: peers defined in sip.conf do not "magically go away". |
| 13:43.58 | [TK]D-Fender | Free99: We have no proof that things are even matching the peer you think it is. |
| 13:44.20 | [TK]D-Fender | Free99: And a new calls takes the CURRENT dialplan. There is no "area". That scope does not exist. |
| 13:44.21 | Free99 | igcewieling, I'm not saying they are "going away" |
| 13:44.35 | [TK]D-Fender | Free99: There is no little microcosm tied to a "peer" |
| 13:44.43 | Free99 | [TK]D-Fender, I'm saying that the peer fails to be confined in the correct context |
| 13:44.53 | igcewieling | sorry registrations, "is there a way to do a reload without losing all my registrations?" |
| 13:45.11 | Free99 | igcewieling, the other peers are realtime |
| 13:45.14 | [TK]D-Fender | Free99: show us. |
| 13:45.22 | Free99 | I didn't know which ones you were asking about, sorry |
| 13:45.50 | igcewieling | Free99: Ah. They is not unusual. It means you screwed up, not that there is a bug |
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| 13:47.26 | Free99 | [TK]D-Fender, gathering data. |
| 13:47.34 | Free99 | igcewieling, care to elaborate? |
| 13:47.58 | igcewieling | Free99: could be one of a dozen or more errors |
| 13:48.32 | igcewieling | have you pastebin'd the peer yet? |
| 13:48.43 | igcewieling | a sip debug of an incoming call not matching the peer? |
| 13:49.47 | Free99 | boss is on my ass, just a second |
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| 14:00.11 | Free99 | do I need to pb the static peer (the termination provider?) their context is set to a2incoming |
| 14:02.41 | [TK]D-Fender | Free99: You need to show use the CALL |
| 14:02.50 | [TK]D-Fender | Stop staring at configs |
| 14:02.58 | [TK]D-Fender | we have no prrof the call is even matching your peer |
| 14:03.16 | [TK]D-Fender | So show the call and show the bits related to it |
| 14:03.58 | Free99 | [TK]D-Fender, I apologize but we're going to have to wait for a bit because I cannot break the system while my boss is doing a demo |
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| 14:04.13 | [TK]D-Fender | I also don't recall being able to use realtime for SOME peers, and the .conf file for others |
| 14:04.39 | igcewieling | [TK]D-Fender: It works. |
| 14:04.50 | igcewieling | I did it before we abandoned realtime |
| 14:04.52 | [TK]D-Fender | If you say so |
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| 14:12.11 | blitzrage | we have static (in sip.conf) and realtime peers at the same time |
| 14:12.23 | blitzrage | static are gateways, realtime peers are devices/phones |
| 14:13.29 | igcewieling | blitzrage: a very handy feature |
| 14:14.00 | file | internally chan_sip looks through its internal list, and then queries realtime |
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| 14:15.35 | igcewieling | Whoo! Whoo! I new something [TK]D-Fender didn't! 8-) |
| 14:15.41 | igcewieling | knew, even |
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| 14:21.21 | carrar | LIES!! |
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| 14:40.22 | jmetro | two things can register to the same sip.conf entry at the same time? |
| 14:40.25 | jmetro | or does that break |
| 14:43.28 | carrar | The devices can, but asterisk will send the call to the last one to register |
| 14:44.48 | jmetro | darn. im trying to lump two phones together that should always get called together even if i just dial the sip peername |
| 14:45.04 | file | unpossible |
| 14:45.13 | carrar | other pbx systems do that |
| 14:45.30 | Greenlight | That breaks the defenition of a peer |
| 14:45.32 | file | yes, it's an implementation detail of chan_sip |
| 14:45.57 | jmetro | its a wireless handset that isnt integrated, it SHOULD be part of the peer |
| 14:46.06 | carrar | It's not hard to work around that with user peers |
| 14:46.12 | carrar | two |
| 14:47.47 | jmetro | carrar: elaborate? |
| 14:48.11 | carrar | so have 5555-desk 5555-soft 5555-cordless |
| 14:48.21 | carrar | check to see if registered before dialing all three |
| 14:48.25 | carrar | from a single extension |
| 14:49.02 | carrar | you would have a extension that dials 3 local extensions |
| 14:49.25 | jmetro | hm |
| 14:49.28 | carrar | which everone answers gets the call |
| 14:49.53 | jmetro | but lets say i add that extension to a queue |
| 14:49.56 | carrar | can add a 4th (cell) with a delay |
| 14:49.58 | jmetro | how does the queue dial it |
| 14:50.05 | Greenlight | Via "Local/EXT" |
| 14:50.19 | carrar | queue uses the peername |
| 14:50.55 | jmetro | because i've done the "one extension rings multiple" before but im trying to do this on a queue member |
| 14:51.06 | carrar | let me make a example for ya |
| 14:53.24 | carrar | https://www.osburn.com/jmetro.txt |
| 14:53.27 | carrar | something like that |
| 14:54.08 | jmetro | yeah, basic workgroup dials, you just stick the & in ther |
| 14:54.24 | jmetro | but extensions dont match peernames, so the queue wouldnt call the other lines |
| 14:54.46 | jmetro | unless youre saying make the peername the extension and that...works somehow |
| 14:55.11 | carrar | most people don't want the queue dialing all their phones |
| 14:55.21 | carrar | least our customers anyways |
| 14:55.25 | jmetro | =D my call center agent does. |
| 14:55.28 | carrar | YMMV |
| 14:55.53 | carrar | when they login to the queue add all their devices |
| 14:56.24 | jmetro | hm, then my interface that shows queue members will be flooded |
| 14:57.10 | carrar | Sounds like you need to have a meeting with your call center manager |
| 14:57.24 | carrar | and write some policies :) |
| 14:58.16 | jmetro | well, that would be my boss and i dont think he will like being told that my agent cant get queue calls on the mobile <.< |
| 14:58.29 | carrar | You coulg go as far as writting a web page app that lets people select what devices join the queue |
| 14:58.37 | carrar | when they login |
| 14:58.45 | carrar | then check that db when joining a queue |
| 15:00.02 | *** join/#asterisk roma (~roma@67-42-129-46.albq.qwest.net) |
| 15:00.16 | pabelanger | that's what we did :) |
| 15:01.20 | carrar | nice, there just hire some Chineese hackers to steals pabelanger dialplan and configs |
| 15:01.34 | carrar | heh |
| 15:02.14 | pabelanger | carrar: just go to astricon, presenting something there |
| 15:03.19 | carrar | nice |
| 15:03.26 | carrar | posting your code too? |
| 15:03.46 | carrar | jmetro, there you go, company reason to attend astricon |
| 15:04.38 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 15:04.40 | jmetro | we already go =p |
| 15:06.00 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 15:10.32 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.141) |
| 15:11.20 | *** join/#asterisk pet4153 (~pet@adsl-84-226-12-13.adslplus.ch) |
| 15:12.34 | pet4153 | ? |
| 15:12.44 | Chainsaw | Yes? |
| 15:13.18 | WIMPy | No. |
| 15:13.21 | Greenlight | 42 |
| 15:13.30 | pet4153 | hi there i might need some help to set up a conference bridge with the webgui |
| 15:13.42 | Chainsaw | Ah, FreePBX? |
| 15:13.50 | WIMPy | Which one? |
| 15:14.19 | pet4153 | yes running on a synology v1.8 webgiu 2.1 |
| 15:14.40 | WIMPy | What the heck is that? |
| 15:15.12 | jmetro | <PROTECTED> |
| 15:15.18 | Chainsaw | jmetro: You are correct. |
| 15:15.24 | pet4153 | hiii syno is as common nas system comes with a lot of packages like |
| 15:15.50 | pet4153 | ok try to go on freepbx |
| 15:16.22 | pabelanger | carrar: some code |
| 15:16.53 | Chainsaw | pabelanger: Not the telemarketer maze? |
| 15:17.35 | malcolmd | i think those sinology things use the old asterisk-gui |
| 15:17.36 | carrar | synology's are great |
| 15:17.42 | carrar | we have 2 |
| 15:17.57 | WIMPy | I don;t see any reference to Asterisk at Synology. |
| 15:18.07 | carrar | I have one here at home for my personal NAS |
| 15:18.14 | malcolmd | http://www.synology.com/releaseNote_enu/package_Asterisk.php?lang=enu |
| 15:18.25 | pabelanger | Chainsaw: huh? |
| 15:18.41 | malcolmd | WIMPy: they threw asterisk on it, then promptly sent everyone of their users to the asterisk forums for support when they had questions. hooray synology |
| 15:18.50 | carrar | hahah |
| 15:18.54 | carrar | nice |
| 15:19.03 | carrar | I'll help em |
| 15:19.24 | carrar | gets his book link ready and his compile form source remarks ready |
| 15:19.36 | malcolmd | yeah, it's been an awesome experience for the users and for the forums responders. and by awesome, i mean "not awesome." i really wish they'd get more engaged in helping their users. |
| 15:20.28 | pet4153 | well i got everything to work very quick except conferences... |
| 15:20.47 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.141) |
| 15:21.45 | WIMPy | pet4153: To use ConfBridge, you want a newer version of Asterisk anyway. |
| 15:22.10 | pet4153 | ok... |
| 15:30.52 | *** join/#asterisk pet4153 (~pet@adsl-84-226-12-13.adslplus.ch) |
| 15:43.01 | malcolmd | and asterisk-gui was never updated to use confbridge |
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| 16:28.17 | DBordello | Are there any web guis worth using for monitoring, etc? I still want to do the configuration using the config files. |
| 16:28.22 | DBordello | FreePBX doesn't seem like the right solution |
| 16:28.35 | Greenlight | We use zabbix |
| 16:28.42 | Greenlight | Via SNMP |
| 16:28.43 | WIMPy | What do you want to monitor? |
| 16:29.03 | *** join/#asterisk navaismo (~navaismo@189.241.9.57) |
| 16:29.03 | DBordello | WIMPy, Not sure. Phones registered, etc. Statistics are fun :) |
| 16:29.13 | DBordello | Greenlight, interesting. |
| 16:29.16 | *** join/#asterisk thehar (thehar@diddlebox.thehar.com) |
| 16:29.20 | igcewieling | DBordello: monitoring? FOP2 is what we use, but I'm not involved in installing, maintaining, or supporting it. |
| 16:29.35 | Greenlight | We use it to measure number of active channels of each type and graph it to big TV's on the wall :) |
| 16:29.49 | DBordello | Greenlight, that kind of fun stuff :) |
| 16:30.12 | DBordello | igcewieling, that looks cool |
| 16:30.16 | Greenlight | Yup -- looks very techy :) |
| 16:31.04 | DBordello | Looks like overkill for my purposes :) |
| 16:31.17 | WIMPy | The only thin I managed to do via SNMP to Asterisk was to make it crash. |
| 16:31.24 | Greenlight | Heh |
| 16:31.43 | Greenlight | I do remember it beign a royal pain to setup, but once we got it going it's worked flawlessly |
| 16:31.59 | Greenlight | We monitor a dozen or so asterisk boxes from our main zabbix |
| 16:32.27 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 16:32.44 | Greenlight | And automated email alerts tied in based on certain criteria as well |
| 16:32.49 | Greenlight | All good stuff |
| 16:33.13 | DBordello | This is a home based PBX, nothing fancy :) |
| 16:33.45 | DBordello | Installing Fop2 now |
| 16:34.34 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 16:37.30 | DBordello | Actually, that is probably a bit overkill. I think i'll stick with sip show peers |
| 16:37.48 | danfromuk | Hi. Using AMI, whats the easiest way to find out whether a SIP Peer is inuse? |
| 16:38.14 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 16:38.52 | Greenlight | The ExtensionState action I'd reckon |
| 16:39.04 | WIMPy | Use the DEVICE_STATE function with GetVar. |
| 16:39.21 | WIMPy | Or if you're connected anyway, listen for the events. |
| 16:39.40 | danfromuk | Greenlight: I saw that, but that would require me to set up hints where I currently dont have. |
| 16:39.47 | DBordello | I have an incoming route that I would like to ring multiple phones (this is at home). Is there a better way to do this than for the incoming context using Dial(SIP/phone1&SIP/phone2....)? This seems clunky and requires reconfiguration if I add a new phone. Is it possible to group them? |
| 16:40.07 | danfromuk | WIMPy: DEVICE_STATE looks like it would work. I need to get an initial status, then I can monitor for events. |
| 16:40.25 | danfromuk | Thanks for both of you. |
| 16:40.46 | WIMPy | DBordello: Exactely the way you said. |
| 16:41.04 | WIMPy | Or you can define a global variable, but I'm not sure that makes things cleaner. |
| 16:41.18 | DBordello | WIMPy, just concat'ing them in the Dial(..)? |
| 16:41.34 | *** join/#asterisk imox (~imox@24.134.18.71) |
| 16:41.35 | WIMPy | Err, what? |
| 16:41.58 | DBordello | <PROTECTED> |
| 16:42.07 | Greenlight | DBordello: The only other option that springs to mind is to use a Queue with RingAll |
| 16:42.12 | WIMPy | yes |
| 16:42.48 | tech_travis | DBordello: what about trying SLA (shared line appearance)? |
| 16:42.49 | DBordello | Okay, sounds good |
| 16:43.39 | DBordello | tech_travis, that sounds promising. I am really kind of looking for a traditional home phone system, where each phone has several lines, tied to outgoing routes. All will ring, etc. |
| 16:44.15 | *** join/#asterisk jhirley (~chatzilla@c-75-74-4-9.hsd1.fl.comcast.net) |
| 16:45.09 | DBordello | tech_travis, that looks exactly like what I want, thanks for the tip |
| 16:45.19 | tech_travis | DBordello: np. |
| 17:12.11 | danfromuk | WIMPy: DEVICE_STATE seems to always return NOT_INUSE |
| 17:12.36 | WIMPy | Do you hace call counters enabled? |
| 17:12.40 | WIMPy | have |
| 17:13.05 | danfromuk | I thought so. |
| 17:13.11 | danfromuk | One moment. |
| 17:15.36 | *** join/#asterisk m0spf (~steve@2001:ba8:1f1:f12e::2) |
| 17:16.00 | danfromuk | Which variable enabled call counters? limitonpeers ? |
| 17:16.21 | WIMPy | dpends on the version. |
| 17:16.30 | danfromuk | Its 1.8 |
| 17:17.17 | WIMPy | is not good with the sip stuff. |
| 17:17.43 | danfromuk | Got it |
| 17:19.40 | blitzrage | callcounter=yes I think |
| 17:20.21 | blitzrage | validates |
| 17:20.30 | blitzrage | ya, it's callcounter=yes in at least 1.8+ |
| 17:20.58 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 17:21.37 | *** join/#asterisk vlad_starkov (~vlad_star@109.95.84.114) |
| 17:23.18 | danfromuk | Perfect. Its working now. Does callcounter=yes affect anything apart from DEVICE_STATE? eg. can it prevent calls from being accepted if the device already has one call active? |
| 17:23.53 | WIMPy | Not unles you also set a limit. |
| 17:24.49 | danfromuk | If callcounter=no and call-limit=1, does the limit not work? |
| 17:25.12 | WIMPy | I don't think so. |
| 17:25.40 | Qwell | callcounter=yes sets call-limit to a very high number. |
| 17:26.22 | Qwell | They act on the same object. |
| 17:26.43 | danfromuk | Ok, thanks. |
| 17:31.44 | danfromuk | Sorry, final question, whats the maximum value for actionid in the AMI? |
| 17:33.09 | Qwell | It's an arbitrary string. |
| 17:33.51 | [TK]D-Fender | I wonder why it wouldn't always just count... |
| 17:35.05 | *** join/#asterisk Tarso (~Tarso@189.61.52.46) |
| 17:40.53 | *** join/#asterisk _omer (omer@184.175.79.212) |
| 17:40.59 | _omer | is there any GUI for Asterisk 11 ? |
| 17:42.39 | danfromuk | Qwell: thanks |
| 17:43.30 | blitzrage | _omer: AsteriskNOW 3.0.x uses FreePBX and Asterisk 11 base |
| 17:43.40 | navaismo | _omer, Most used FreePBX |
| 17:43.52 | *** join/#asterisk serafie1 (~erin@user-24-214-173-250.knology.net) |
| 17:45.46 | *** join/#asterisk dtascom (~dtascom@cpe-142-129-156-205.socal.res.rr.com) |
| 17:52.30 | [TK]D-Fender | Which of course... he's already using... |
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| 18:09.22 | *** join/#asterisk AkkerKid (~AkkerKid@23.31.20.201) |
| 18:09.32 | AkkerKid | howdy all!! |
| 18:11.36 | AkkerKid | Does anyone know of a pre-existing network of people and asterisk boxen that share VOIP services worldwide? I would like to trade one of my US Sip trunks for a German one for a while... |
| 18:13.49 | jmetro | i trade european numbers for US numbers but thats a limited exchange |
| 18:15.13 | navaismo | kamalio can be used as SBC? |
| 18:15.25 | AkkerKid | I'm justthinking I'd love for my family to be able to call their german friends back home without a high phone bill |
| 18:15.34 | *** join/#asterisk alagar (~helpdesk@vsusg1.vernalissystems.com) |
| 18:15.36 | igcewieling | For the most part, if you have low volume usage it is not worth the hassle, calls to Europe are so cheap and nobody wants to exchange large volumes of traffic |
| 18:15.53 | AkkerKid | hmmm... ok |
| 18:16.12 | WIMPy | AkkerKid: Why don;t you just get an account at the destination? |
| 18:16.21 | AkkerKid | that was my next question. |
| 18:16.43 | AkkerKid | are there any decent providers that would sip trunk EU numbers to me in the USA? |
| 18:16.51 | AkkerKid | cheaply enough? |
| 18:17.22 | WIMPy | For germany you will only get service numbers unless you can get a post box somewhere. |
| 18:17.37 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 18:17.54 | AkkerKid | maybe I should drop a GSM gateway on some relative's internet connection... :) |
| 18:18.03 | AkkerKid | get cell service |
| 18:18.19 | WIMPy | That might be a little expensive. |
| 18:18.55 | WIMPy | (and senseless) |
| 18:19.41 | WIMPy | Do you want both ways? |
| 18:20.15 | AkkerKid | for about $150US I could get a gsm gateway and find a cheap prepaid cell service provider... |
| 18:20.50 | WIMPy | That wold still cost you twice as much than a sip account. |
| 18:21.03 | AkkerKid | i suppose. |
| 18:21.14 | *** join/#asterisk m0spf (~steve@2001:ba8:1f1:f12e::2) |
| 18:21.15 | AkkerKid | i would want both ways though |
| 18:22.12 | WIMPy | You need someone to get you a number unofficially then. |
| 18:28.08 | *** join/#asterisk mrothe (~mrothe@exherbo/developer/pdpc.active.mrothe) |
| 18:28.25 | navaismo | Any Ideas on how to hide/encrypt/license a part of dialplan?? |
| 18:29.16 | [TK]D-Fender | Do your work in AGI and encrypt that. |
| 18:29.17 | WIMPy | AGI |
| 18:29.21 | [TK]D-Fender | That's all. |
| 18:29.25 | mrothe | Hello. I want to be reachable by sip:me@domain. can I use asterisk as a sip server for this? or is this a job for kamailio? |
| 18:29.40 | [TK]D-Fender | mrothe: Yes you can do this with * |
| 18:29.43 | WIMPy | yes and yes |
| 18:29.56 | mrothe | [TK]D-Fender: okay. thanks. I'll look into it then. |
| 18:30.25 | [TK]D-Fender | mrothe: It'll be an anonymous SIP call and targets exten =>me |
| 18:30.27 | navaismo | AGI? still plain text, and use for a complete call its a bad practice . |
| 18:30.50 | mrothe | kamailio has a *horrible* build system, so I'd prefer asterisk. |
| 18:30.59 | [TK]D-Fender | navaismo: ENCRYPT the AGI |
| 18:31.00 | mrothe | [TK]D-Fender: thank you. I'll try. :-) |
| 18:31.02 | WIMPy | Whay does AGi have to be plain text? |
| 18:31.16 | [TK]D-Fender | It doesn't |
| 18:33.16 | navaismo | ok besides the encrypt stuff an AGI for all , ill check it |
| 18:37.57 | *** join/#asterisk alagar (~helpdesk@vsusg1.vernalissystems.com) |
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| 18:52.01 | *** join/#asterisk [404] (~404]@12.179.117.114) |
| 18:54.40 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.140) |
| 18:54.46 | [404] | I was wondering if someone could point me in the right direction if this can even be done. I am trying to setup a voicemail where depending on the extension and certain variables, I would like the files to be saved under those directories. lets say I have 2 facilities and 2 people in each facility, I would like the voicemails to be saved int parent_directory/facility/person/*.wav can this be done? |
| 18:55.35 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.125.182) |
| 18:55.43 | [TK]D-Fender | [404]: Symlink the folders |
| 18:55.44 | WIMPy | That's the way it's done if you use contexts. |
| 18:55.53 | WIMPy | If you don't mind the extra INBOX/. |
| 18:57.04 | [404] | well, I am going to have over 200 facilities that are pulled from db and there could be over 10000 people per facility |
| 18:58.35 | [TK]D-Fender | how is a "facility" pulled down from a DB? |
| 18:58.46 | [404] | from an ID |
| 18:59.08 | [404] | agi script if thats what you were looking for |
| 18:59.36 | [TK]D-Fender | That doesn't really mean much to us. Anyway... files are files.... voicemail is stored under the varspool folder defined in asterisk.conf |
| 18:59.48 | [TK]D-Fender | if you want that to go somewhere else per-user, then manually symlink it. |
| 18:59.51 | WIMPy | That sounds like you don;t intend to use voicemail, but build somethign yourself. |
| 19:00.21 | igcewieling | [404]: use voicemail contexts |
| 19:03.06 | [404] | k, i will check it out |
| 19:17.40 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:24.45 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 19:24.45 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:27.56 | *** join/#asterisk picard276 (~slester@64-79-127-110.static.wiline.com) |
| 19:30.18 | picard276 | hey guys im having a weird sip issue |
| 19:30.28 | picard276 | dialing from Exten 35 to 30 works |
| 19:30.35 | picard276 | but dialign from exten 30 to 35 does not work: |
| 19:30.40 | picard276 | here is the sip logs |
| 19:30.44 | picard276 | http://pastebin.com/v8WewZVu |
| 19:30.57 | pabelanger | sounds like a registration issue |
| 19:31.28 | [TK]D-Fender | Found peer '30' for '30' from 188.165.231.30:12060 <--- Reliably Transmitting (NAT) to 188.165.231.30:12060 ---> SIP/2.0 403 Forbidden |
| 19:31.40 | [TK]D-Fender | 35 is not the problem. |
| 19:31.46 | [TK]D-Fender | Nor is your dialplan. |
| 19:31.50 | [TK]D-Fender | The peer did not auth |
| 19:33.54 | picard276 | 30 did not auth you mean? |
| 19:34.03 | [TK]D-Fender | Very clearly. |
| 19:34.13 | [TK]D-Fender | The giant "FORBIDDEN" sign tipped me off |
| 19:35.09 | picard276 | kk ill retake a look at that |
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| 19:46.41 | *** part/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
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| 19:56.09 | mtnbkr | hello everyone... I have been looking into the dahdi_* command to see if there was something I could use to test whether or not all 4 of our POTS lines were connected to the CO. We have had issues with recent thunderstorms and tornado warnings where there was no voltage on one or more of the lines and I'd like to write a custom test for our monitoring software (Xymon) so that we are notified immediately, and reminded unti |
| 19:56.09 | mtnbkr | l th eissue is reasolved. Was hoping for a quick command line prog that would report the status of the ports which could be parsed in a script. Thanks! |
| 20:00.32 | DBordello | Does aterisk have a native since of a phone book? |
| 20:00.37 | DBordello | sense* |
| 20:01.19 | DBordello | that can be pushed to the phones |
| 20:02.29 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 20:03.09 | [TK]D-Fender | DBordello Nope. |
| 20:03.37 | [TK]D-Fender | DBordello: My * is a jukebox and coffee timer. What are these "phone" things you're referring to? |
| 20:04.11 | DBordello | [TK]D-Fender, desk paper weights |
| 20:04.42 | [TK]D-Fender | You still use dead trees for communication? How quaint.... |
| 20:05.37 | mtnbkr | also, dahdi_tool (txt "gui") correctly shows me what I am looking to monitor "Total/Conf/Active" but it interactive only. Am I looking in the wrong place? |
| 20:09.06 | *** join/#asterisk myk_ (~mizan@180.234.138.23) |
| 20:11.54 | mtnbkr | Hmmm I think I am getting warmer... lol asterisk -rx "dahdi show channel x" grep for InAlarm field. thanks! |
| 20:13.55 | igcewieling | mtnbkr: "cat /proc/dahdi/1" |
| 20:14.57 | mtnbkr | igcewieling: Ah that works too grep "RED" and done. :) Thanks |
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| 20:41.12 | picard276 | TK im still having that issue |
| 20:41.18 | picard276 | how is it forbidden |
| 20:41.24 | picard276 | if i can call from 35 to 30 |
| 20:41.31 | picard276 | how is 30 forbidden it must be registered for that to happen |
| 20:41.41 | picard276 | if i do sip show peers i can see 30 registered at the right IP address |
| 20:41.44 | igcewieling | picard276: incorrect. |
| 20:41.47 | Qwell | Registration has nothing to do with making a call. |
| 20:42.03 | igcewieling | A peer does not have to be registered to make a call, only to receive a call (and even then not in all setups) |
| 20:42.11 | picard276 | right that is my point |
| 20:42.17 | picard276 | so the peer can recieve a call just fine |
| 20:42.20 | picard276 | it just can't make a call |
| 20:42.35 | picard276 | ill post the logs one sec |
| 20:42.38 | Qwell | Correct. The two have nothing to do with each other. |
| 20:44.37 | Free99 | anyone know how to enforce a jitter buffer in both directions? Link goes through cruddy satellite connection, force enabled JB but it seems to only work in one direction |
| 20:46.44 | igcewieling | Free99: audio must be dejittered on the endpoints. Asterisk's jitter buffer is used for DAHDI and MeetMe/ConfBridge/Voicemail/etc. i.e. stuff where Asterisk is an endpoint. |
| 20:47.35 | igcewieling | if you have directmedia enabled the audio doesn't even pass through Asterisk |
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| 20:48.14 | Free99 | igcewieling, can't seem to find that setting for Linksys pap2t |
| 20:49.43 | igcewieling | Free99: then you have not been putting the right words like "Linksys pap2t jitter" into google |
| 20:51.19 | jmetro | anyone know a way to end the autodestruct spam? |
| 20:51.53 | igcewieling | jmetro: directmedia=no |
| 20:52.43 | jmetro | 3: |
| 20:53.45 | igcewieling | jmetro: that is the way. You might also reports it as a bug on Jira. The bug is that the message level should not be a warning, as it does not indicate an error or problem |
| 20:54.17 | jmetro | you would think asterisk could just ... stop banging its head on the wall |
| 20:54.29 | picard276 | http://pastebin.com/NNmN1qvz |
| 20:54.39 | picard276 | there are my sip logs |
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| 20:56.40 | igcewieling | picard276: did you fail to mention you are using webrtc or did I miss it? |
| 20:56.50 | igcewieling | use a real softphone and see if it works |
| 20:58.41 | igcewieling | "SIP/2.0 488 Not acceptable here" <-- you have a codec issue |
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| 21:08.09 | Free99 | igcewieling, thanks for the obvious tip |
| 21:08.48 | jmetro | codec issues get me all the time when im working with new systems |
| 21:09.22 | igcewieling | jmetro: I never have a problem with codecs. ulaw, g729 and g722 are all we allow |
| 21:12.13 | mtnbkr | igcewieling: thanks again for the help, and have a great weekend! |
| 21:12.22 | *** part/#asterisk mtnbkr (~mtnbkr@75-150-91-17-NewEngland.hfc.comcastbusiness.net) |
| 21:13.10 | jmetro | igcewieling: same here, but we lock it down to only g722 or only g711 |
| 21:13.53 | igcewieling | we have close to 500 channels of g729 transcoding capacity so we use it a lot, also our carriers send calls to us as g729 |
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| 21:25.09 | avb | hello, im trying to send a message 5060/udp from php, asterisk replying me according to cli, but my script most of the time 'losing' the reply. could be anybody was making something like that before? |
| 21:26.04 | avb | funny thing is that im getting a replies if im running the script once a minute |
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| 21:26.27 | avb | feels like port getting locked |
| 21:26.55 | [TK]D-Fender | And where is your script running from? |
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| 21:39.13 | Free99 | ok, having trouble with google on this one: I have people dialing with a + in front. How do I filter that plus out so I can match extensions? |
| 21:39.33 | Free99 | googling "remove + asterisk dialplan" was not so useful haha |
| 21:39.40 | avb | [TK]D-Fender: what do you mean? |
| 21:39.57 | avb | [TK]D-Fender: im connecting remotely |
| 21:40.08 | avb | and i have a root on the server site |
| 21:40.19 | avb | so im looking at sip set debug output |
| 21:40.19 | [TK]D-Fender | Free99: Viable basics... trim off the leading + |
| 21:40.44 | [TK]D-Fender | Free99: And clearly you have to have a pattern to match it in the first place so you can even process it |
| 21:41.07 | [TK]D-Fender | avb: What machine is your script running on? |
| 21:41.19 | jmetro | Free99: have a "exten => _+.,1,goto(${EXTEN}) |
| 21:41.44 | [TK]D-Fender | Which will fail... |
| 21:41.51 | [TK]D-Fender | But close to an idea.... |
| 21:42.21 | Free99 | jmetro, there's no way to just remove the +? I can't write into EXTEN obviously |
| 21:42.43 | [TK]D-Fender | FreeYou don't .. you USE only PART of it in your DIAL then |
| 21:43.02 | ChannelZ | ${EXTEN:1} and such |
| 21:43.23 | [TK]D-Fender | Variables 101 |
| 21:43.25 | [TK]D-Fender | ~book |
| 21:43.26 | 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 |
| 21:43.27 | ChannelZ | but if you're picking up the + via a wildcard you need to do that discriminately |
| 21:43.28 | [TK]D-Fender | ^^^^ |
| 21:43.29 | [TK]D-Fender | ^^^^ |
| 21:44.19 | Free99 | ChannelZ, what do you mean by discriminatingly |
| 21:44.44 | Qwell | He didn't say discriminatingly. |
| 21:45.05 | Free99 | qwell, autocorrect |
| 21:45.14 | ChannelZ | IE if your extension is _. (match anything) but what gets dialed may or may not start with +, ${EXTEN:1} will strip off the first digit regardless. |
| 21:45.42 | Free99 | oh ok. I can use a gotoif to determine if it has that |
| 21:46.01 | avb | [TK]D-Fender: linux |
| 21:46.04 | ChannelZ | right-o, or a simple inline-expression probably would work |
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| 21:48.16 | Free99 | ChannelZ, if I wanted to (after removing the +) go to an extension, you think there are any downsides to goto(context1,${filtered_var})? |
| 21:48.34 | Free99 | I basically want to start the matching over again |
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| 21:49.17 | ChannelZ | well garbage-in/garbage-out rules apply as ever |
| 21:49.33 | ChannelZ | but otherwise no you could do that |
| 21:52.18 | Free99 | thanks for your help ChannelZ |
| 21:54.04 | ChannelZ | sure |
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| 22:04.20 | ChannelZ | BTW $["${EXTEN:0:1}"="+"?${EXTEN:1}::${EXTEN}] should return the extension stripping off a + if it exists |
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| 22:12.06 | igcewieling | or you could use exten => _+1NXXNXXXXXX,1,Goto(${EXTEN:1},1) |
| 22:12.58 | igcewieling | what, you thought extensions could only contain 0-9, N, X, Z, and . ? |
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| 22:16.23 | ChannelZ | except it's a "sometimes I get +, sometimes I don't" |
| 22:16.34 | ChannelZ | but either way |
| 22:16.56 | ChannelZ | wasn't clear how many different extensions there might be that could be dialed that way (locals, internals, etc) |
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| 22:22.19 | Free99 | a question for you: I'm setting up the rtt value on a device, can I use the value from "sip show peers" regarding the number of milliseconds? |
| 22:22.43 | Free99 | satellite is terrible, it's like 800ms for all peers average |
| 22:25.12 | sweeper | Free99: you have terrible satellite then ;) |
| 22:25.32 | sweeper | the only 'unavoidable' latency is ~480ms |
| 22:25.48 | Free99 | at $6500 per mbps, you know why lol |
| 22:26.05 | sweeper | well yo're also getting screwed on bandwidth then o.O |
| 22:26.21 | sweeper | we charged about ~3k/mbps |
| 22:26.33 | Free99 | really? where's your teleport? |
| 22:26.36 | sweeper | you got some crappy c-band or something? |
| 22:26.39 | sweeper | singapore |
| 22:26.57 | Free99 | yeah, it's C b/c we're making a dish for entry level marine companies |
| 22:26.59 | sweeper | have dealer agreements with some service out of MD and HK |
| 22:27.12 | sweeper | ewewew |
| 22:27.16 | sweeper | run your numbers man |
| 22:27.29 | Free99 | dude, I'm not in charge of that lol |
| 22:27.46 | Free99 | but if you have anything you can show me, I can take it to the people who matter |
| 22:29.34 | sweeper | well I don't work there anymore, but have your guys ask for a quote from jarel@jpiworldwide.com, I'm sure as hell he'll do better than $6k |
| 22:30.05 | sweeper | also I was in charge of putting in a new set of infrastructure, with some super awesome new modems that are TINY |
| 22:30.16 | sweeper | if you want to keep in cheap, def check it out |
| 22:30.30 | sweeper | http://www.romantis.com/ <-- we use that stuff |
| 22:31.18 | sweeper | that modem is the size of your hand |
| 22:31.26 | sweeper | and does SCPC at 16APSK |
| 22:32.02 | sweeper | remote<->teleport latency is 550ms, with about 10ms jitter |
| 22:33.16 | Free99 | we use iDirect right now |
| 22:33.22 | sweeper | ahahahahaha |
| 22:33.34 | sweeper | ok yea, DEFINITELY email JP |
| 22:33.56 | sweeper | romantis modem costs about 50% and is so much cooler |
| 22:34.04 | Free99 | let's see what happens |
| 22:34.08 | sweeper | and SCREW TDMA if you're doing voice |
| 22:34.39 | Free99 | thanks for the tip sweeper. I'll do some reading |
| 22:34.43 | sweeper | n/p |
| 22:34.52 | sweeper | you could also go the DIY oute |
| 22:34.54 | sweeper | *route |
| 22:35.07 | Free99 | which it looks like we're going to be doing tbh lol |
| 22:35.22 | sweeper | romantis hub is very cheap, our singapore hub does 40 remotes, only cost $10k all in |
| 22:35.26 | Free99 | this idirect stuff has been a bear besides my fumbling with the asterisk system |
| 22:35.34 | sweeper | yeap |
| 22:36.03 | sweeper | hey, you could stick an rpi, and a romantis, and a mikrotik, in a box, bam, 1U pbx+router+vsat |
| 22:36.07 | sweeper | you doing stabilized? |
| 22:38.41 | sweeper | if you talk to romantis, tell them Aleks Clark sent yu and said to give you the setup he bought ;) |
| 22:40.32 | sweeper | oh and vagan will probably offer you teleport services, but usually he's way over priced |
| 22:41.45 | sweeper | Free99: if you have any vsat or vsat <-> voip questions, feel free to shoot me an email, aleks.clark@gmail.com |
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