| 00:18.30 | *** part/#asterisk rue_mohr (~rue@h24-207-19-104.cst.dccnet.com) |
| 00:23.39 | *** join/#asterisk JuStIcIa_ (~justicia@190.80.137.167) |
| 00:24.14 | carrar | peeps out |
| 00:24.25 | carrar | err perhaps in |
| 00:24.49 | carrar | looks at Katty |
| 00:29.55 | *** part/#asterisk msuszczyn (~msuszczyj@89.pool85-60-20.dynamic.orange.es) |
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| 00:52.33 | *** join/#asterisk jasoncarter (5006fcbf@gateway/web/freenode/ip.80.6.252.191) |
| 00:53.44 | jasoncarter | Can asterisk be made to load a file or send an email once a caller has disconnected from the server? |
| 01:08.36 | jc319 | This is my first Asterisk dialplan experience, everthing was going great so far with loads of help from p3nguin & wiretap but when it came to voicemail I think I have either a minor design flaw or a lack of knowledge how to fix this >> In my design there are two users (user1, user2), each have access to inbound (DID) and outbound (SIP) phone service. They can access the system on deskphone |
| 01:08.36 | jc319 | & softphonePC & softphoneMobile. They might be 'online' at the same time on more than one device (e.g. @home therefore has access to deskphone in theory but at some moment can be in the living room/garden, therefore softphoneMobile in 'online' too). They have 2 DID #s each (say work# & personal#). to be able to distinguish different endpoints, each endpoint (desk/pc/mobile) is defined as a |
| 01:08.36 | jc319 | different sip peer for both of them. |
| 01:08.56 | jc319 | How can I give them access to the same mailbox? say user1 has 201 extension for deskphone, 202 for PC and 203 for mobile. the other user has 211,212 and 213 respectively for the same devices. their mailboxes are defined as device1=201@default , device2=201@default, device3=201@default (since they are the same user no matter what endpoint). |
| 01:09.07 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 01:09.18 | jc319 | First of all is this the correct method to define same person's different endpoint mailboxes? And if it is, my next issue is the deskphone peer has been named as [MAC], therefore it cannot jump direct into the VoiceMailMain() using num variable. Any solutions to this? |
| 01:10.56 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 01:10.57 | jc319 | "DID mailboxes" is a better term probably. (rather than "endpoint mailboxes" above) |
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| 01:42.29 | *** join/#asterisk scalex000 (~chatzilla@190.166.86.72) |
| 01:42.47 | scalex000 | hello, who have exp with swift, how to install |
| 01:42.52 | scalex000 | ? |
| 01:46.31 | *** join/#asterisk thedavidfactor (~david@nc-71-52-232-56.dhcp.embarqhsd.net) |
| 01:47.36 | scalex000 | where I can download swift for asterisk 1.6.0 |
| 01:47.50 | scalex000 | I downloaded one but I get error when make |
| 01:49.18 | cj | hey carrar! |
| 01:49.21 | cj | how's things? |
| 01:49.55 | cj | carrar: I had lunch with Randy B. yesterday. he's still a character :) |
| 01:52.19 | *** join/#asterisk scalex000 (~chatzilla@186.6.7.235) |
| 02:05.57 | scalex000 | hello |
| 02:06.07 | scalex000 | anyone here |
| 02:06.53 | cj | not unless there's a need |
| 02:07.33 | Haraken | does anyone have a moment to help me trouble shoot my message waiting indicator? I tried the common fixes found using google, such as setting pollmailboxes to yes, and setting my pollfreq however these do not seem to fix the problem |
| 02:07.59 | cj | Haraken: have you looked at ~thebook? |
| 02:08.12 | Haraken | which book? |
| 02:08.19 | cj | ~thebook |
| 02:08.19 | 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 02:08.26 | Haraken | yes |
| 02:08.43 | Haraken | I read over the whole voicemail section, most I could tell is that it should work with the options I have set |
| 02:08.47 | cj | I don't have any experience with that topic, yet, so all I can do is recommend references I've found helpful :) |
| 02:08.54 | Haraken | ahh |
| 02:09.01 | cj | are you running 1.8? |
| 02:09.09 | Haraken | yes, I'm wondering if I should try an older version |
| 02:09.18 | cj | I found that the instructions in the book didn't match up with my experiences... then I realized it was documenting 1.8 :) |
| 02:09.33 | cj | are you reading the 3rd edition or a previous edition? |
| 02:09.37 | Haraken | 3rd |
| 02:10.19 | cj | I guess you could fall back to 1.6 and find an older edition of the asterisk book. *shrug* |
| 02:10.24 | cj | or you could read through the code |
| 02:10.51 | cj | I ended up stepping through the res_crypto.c code with gdb today to find out that the debian packages use /usr/share/asterisk/keys instead of /var/lib/asterisk/keys |
| 02:11.01 | jc319 | Haraken: I'm new to Asterisk but as it happens, I have been reading about voicemail for the last 4 hours, if you share your config I can have a look |
| 02:11.04 | Haraken | weird thing is, message waiting indicator works partially. problem is, it only updates when i restart asterisk |
| 02:11.20 | Haraken | jc319, voicemail.conf? |
| 02:12.03 | jc319 | BTW there is such a post which suggests it was a bug in software (in that case) http://www.fonality.com/trixbox/forums/vendor-forums-certified/polycom/mwi-indicator-only-updates-upon-reboot-ip650 |
| 02:12.44 | jc319 | Haraken: voicemail.conf, voicemail section of exten, mailbox section of peer definition and if relevant phones voicemail section |
| 02:13.26 | jc319 | What phones are you using? |
| 02:13.38 | Haraken | cisco 7940 |
| 02:14.47 | scalex000 | can someone help me I try to found a swift application to install |
| 02:14.54 | scalex000 | but I found app with error |
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| 02:15.17 | jc319 | Did you try on a test peer leaving a message and when the MWI is on changing that peers mailbox to something else and sip reload, see if it clears the lamp instantly |
| 02:18.35 | Haraken | Scorpio2007, I'm copying my config to a notepad for pastebin, noe sec |
| 02:18.44 | Haraken | err |
| 02:18.50 | Haraken | that was to you jc319 |
| 02:19.15 | Haraken | jc319, could you decribe that in detail about the test peer? I haven't tried that yet |
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| 02:21.41 | Haraken | jc319, here is my voicemail.conf http://pastebin.com/zcCppLZT |
| 02:22.27 | Haraken | jc319, also did you mean my phone was a test peer or some other device I think that is where the confusion was for me. I've been testing with my phone with different settings and situations |
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| 02:24.29 | jc319 | I think this test might help to see if 7940 is really polling the voicemailbox as it should. Define two peers 201 & 202, both has 'mailbox=201@default' then leave a message to 201. Now 201 & 202 both should have MWI on. (if not due to your original issue, restart asterisk). When you have the lamp on, change 202s sip peer definition to 'mailbox=202@default' (ensure it does not have new voicemail |
| 02:24.29 | jc319 | :) now if you 'sip reload' it should clear the lamp instantly? Does it? |
| 02:25.56 | jc319 | What do you have in peer definitions |
| 02:26.17 | Haraken | jc319, does it matter if 201 and 202 are connected to any devices? |
| 02:26.55 | Haraken | jc319, which file has the peer definitions? |
| 02:28.37 | Haraken | ah I Think i got it |
| 02:28.37 | Haraken | one sec |
| 02:28.40 | jc319 | sip.conf |
| 02:29.30 | jc319 | If you like I can give you my config that works on 7960 so should work on 7940 if your Asterisk has no problems. At least you can narrow down the scope of your troubleshooting. |
| 02:30.28 | *** part/#asterisk jasoncarter (5006fcbf@gateway/web/freenode/ip.80.6.252.191) |
| 02:30.58 | Haraken | that would be good too, I'm pasting my extension info only on pastebin now |
| 02:31.34 | Haraken | please note that ext 204 is the only extension that is supposed to be getting voicemail right now |
| 02:31.48 | Haraken | http://pastebin.com/vqyKPQvN |
| 02:32.36 | Haraken | originally mailbox was set to 204@device, but that wouldn't even get the MWI to light up. at least now as 204@default it lights up when i reload |
| 02:33.35 | Haraken | jc319, as for your last suggestion about seeing if it is polling, I am testing that theory now |
| 02:35.57 | Haraken | I left a message on extension 200 which is not on my cisco 7940, reloaded sip the MWI did not come on |
| 02:37.54 | jc319 | Haraken: What about after restart? |
| 02:38.14 | Haraken | i setup voicemail on the other extension to this phone |
| 02:38.28 | Haraken | left a message and did sip reload. the indicator does come on at that point |
| 02:39.39 | Haraken | restarting asterisk doesn't make it show up |
| 02:40.05 | jc319 | And here is the excerpts from my config, try copying your 3 files to backup. Then edit yours and remove all voicemail related stuff (voicemail exten) from extensions.conf. totally remove voicemail.conf and just paste in what you see. and also edit the new copy of sip.conf carefully to remove any conflicting and non-existant stuff and just the keep ones from mine. If you also set IP/NAT |
| 02:40.05 | jc319 | according to your setup (if necessary) it should work. If this does not work then your Asterisk must be somehow faulty.. http://pastebin.com/ZzhBZC7t |
| 02:42.16 | jc319 | OK a quicker test, rather than the one above >> I don't know if it's necessary but can you remove the extra options in your voicemail.conf, just use this >> 201 => 1234,Namey Surnamey,email@btld -- also add a new extension in the same context rather than having one of them in [other] the other one in [default] |
| 02:42.32 | jc319 | Perhaps it's a context issue... |
| 02:43.15 | jc319 | I gotta go, in worst case try the config I've pasted, if it works you can build from there, testing often, checking when it breaks |
| 02:43.31 | Haraken | jc319, sounds good, thank you for all of your help |
| 02:43.54 | Haraken | I must got as well, about to get off work. will try this when i get home |
| 02:44.04 | jc319 | Good luck |
| 02:49.57 | jc319 | Wow finally could think of renaming 201@default to 001A6CA3696C-PB1@default to jump in direct to the password prompt. VM is finally ready. |
| 03:29.28 | *** join/#asterisk joe4node (4c5825d0@gateway/web/freenode/ip.76.88.37.208) |
| 03:29.39 | joe4node | hello guys |
| 03:30.29 | *** join/#asterisk emsLinux (~dave@190.71.3.255) |
| 03:30.37 | emsLinux | good night everyone |
| 03:31.39 | joe4node | excuse my question if it was too silly: How do I answer an incoming call using Asterisk Manager , I monitor events and get the incoming channel info from their. How can I answer that call and do things with it |
| 03:32.25 | emsLinux | does anyone has some experience with Google Voice? cus i can't make work the inbound calls, this is the output i get when i try calling to the Google Voice number, i dont know what it means, any idea? http://pastebin.us/7050 |
| 03:34.59 | joe4node | there are more than 50 person in here, I do not see any activity. I am not sure if my IRC client is working . IS THERE ANY BODY HOME? does any one see this message |
| 03:35.00 | joe4node | ? |
| 03:36.20 | emsLinux | i do |
| 03:36.28 | joe4node | emsLinux: can u see my messages ? |
| 03:36.35 | joe4node | good , Hi |
| 03:36.53 | joe4node | I am checking the paste bin to see if I can help |
| 03:37.55 | emsLinux | joe4node yes i do |
| 03:38.21 | emsLinux | joe4node thank you, i'm having a headache right now |
| 03:38.25 | joe4node | in the mean while, Do you know how to answer a call using Asterisk Manager API ? |
| 03:40.20 | russellb | joe4node: the AMI is probably not what you want |
| 03:40.41 | russellb | if you want to do direct call control like that, FastAGI is a more appropriate network interface for that |
| 03:40.43 | emsLinux | not really, i'm using FreePBX, and used this tutorial to set the Google Voice account http://michigantelephone.wordpress.com/2010/12/21/how-to-use-google-voice-for-free-calls-on-an-asterisk-1-8freepbx-2-8-system-the-easy-way/ |
| 03:41.25 | emsLinux | I'm not that good yet... |
| 03:43.09 | joe4node | @russellb thank you. you mean i have to send the call to an address:port then do the control from there? does it work like the old DeadAGI where i can still have control after the calling party hangs up ? |
| 03:44.55 | emsLinux | Can anyone help me with the Google Voice problem pls? |
| 03:45.00 | joe4node | @russellb: |
| 03:45.25 | joe4node | emslinux, i am testing it on my box |
| 03:45.53 | russellb | joe4node: yes |
| 03:55.55 | *** part/#asterisk joe4node (4c5825d0@gateway/web/freenode/ip.76.88.37.208) |
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| 04:17.36 | gruvfunk | emsLinux what's the problem? |
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| 04:27.48 | sweni | Does anyone know which unpatched FreePBX php script is exploited by vp_freepbx_exec1 (See voippack1.4)? |
| 04:30.11 | *** join/#asterisk emsLinux (~dave@190.71.3.255) |
| 04:33.06 | emsLinux | Hello, does anyone here has any experience with Google Voice incoming calls? I got this problem, the call gets in and ring the extension, but when i try to answer i only hear silence, and the Google Voice call continue ringing until the Google Voice voicemail system answers. Here is the CLI output when i call to the Google Voice number... http://pastebin.us/7054, Thank you for any help you can give me =) |
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| 04:40.50 | sweni | emsLinux. it looks like the call is being set up fine - the silence probably indicates a problem setting up the RTP stream. To debug this, you may want to switch on sip debugging "sip set debug peer Gtalk" |
| 04:41.19 | turbobee | do you guys prefer using asterisk with a web control panel such as freepbx or by itself? |
| 04:42.32 | sweni | turbobee - I really like FreePBX, and think that pure dialplan is best for dedicated small-function asterisk systems. You can alsways customise FreePBX. |
| 04:44.04 | turbobee | I see, I like freepbx as well but I'm going to be starting fresh again and was debating on not using it to get better acquanted with the config files |
| 04:44.23 | emsLinux | sweni i thinking this could be the problem http://bit.ly/mCmkbL, what do you think? |
| 04:44.56 | turbobee | freepbx was awesome for giving me an idea of how everything is supposed to come together |
| 04:57.49 | sweni | emsLinux - that looks like a workaround to avoid the problem. It would be better to first look at the Sip exchange to see what the two endpoints try to set up. Do you have any other external Sip proviers that are working successfully? |
| 04:58.33 | emsLinux | no i don't |
| 04:59.01 | emsLinux | i dont know if the extension counts, every extension in the server is conected by SIP throught Internet |
| 04:59.07 | emsLinux | and everything is working fine |
| 04:59.30 | sweni | I presume your Asterisk system is behind a NAT? |
| 05:01.53 | sweni | turbobee - it is great to set up one small asterisk system with all your own dialplan, just for education - it will help you understand how you can extend FreePBX. |
| 05:14.54 | emsLinux | sweni it is, i already reconfigure the nat options using FreePBX, everything work ok, but the problem with GV persist |
| 05:17.26 | sweni | emsLinux - try the Sip debug - I am not sure if I gave you the correct peer name - you can also "sip set debug ip x.x.x.x" with the GV ip address if you know it. |
| 05:21.08 | emsLinux | ill try |
| 05:23.52 | MDesade | hello all |
| 05:24.20 | MDesade | i could use some help with dahdi, if anyone here knows about the digium TDM410 cards? |
| 05:28.59 | *** part/#asterisk emsLinux (~dave@190.71.3.255) |
| 05:34.17 | *** join/#asterisk Wiretap7 (~Wiretap@unaffiliated/wiretap) |
| 05:38.40 | MDesade | dun dun dun!! ok, after reading my Dmesg | less, there is a blurb about "wcopenpci [00] Firmware revision 0 not supported by this driver contact voicetronix to have it updated... has ANYBODY here ever flashed their TMD410 cards??? or do i have to contact digium, and get exchanged? |
| 05:51.29 | MDesade | reading digium's site, says the EOL'd the TDM400's and will exchange them for 410P's... guess ill be calling them monday morning?... good to know? |
| 05:59.35 | MDesade | here is my Dmesg output at boot, regarding my firmware on my TDM cards |
| 05:59.39 | MDesade | http://pastebin.com/MLL8W6x4 |
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| 07:34.17 | DND | hi guys |
| 07:34.38 | DND | anyone here has worked with cisco 2901 as the isdn modem? |
| 07:35.37 | DND | we are having problems making outgoing calls using this. we tried the asterisk machine on another office but with a different isdn modem (elcon is the brand) and its working |
| 07:41.37 | WIMPy | s/modem/gateway/ ? |
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| 09:15.13 | Haraken | mzb, something in the svn release of asterisk doesn't agree with mwi |
| 09:15.42 | Haraken | installed stable release and i now have an indicator light :D |
| 09:18.35 | WIMPy | I haven't really looked, but I don;t seem to see old messages any more. |
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| 11:09.44 | ujjain | Hi, I can perfectly use X-liet with Astesrisk, but PAP2T stopped working since last week, I am not sure what changed |
| 11:09.54 | ujjain | PAP2T -> CRegistration state: Can't connect to login server |
| 11:10.16 | ujjain | The Asterisk-server logs registrations for X-Lite, but PAP2T does not seem to reach it, but Hook state is On. |
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| 13:17.24 | scalex000 | good morning I need a good app_swift to installl my Asterisk |
| 13:23.54 | scalex000 | forget it |
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| 13:49.31 | drcode | hi all |
| 13:49.44 | drcode | any one know diastar by dialogic? |
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| 14:38.24 | DND | hi guys, anyone knows why the number is ringing but asterisk is not logging it? |
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| 14:51.27 | wdoekes2 | DND: logger.conf? |
| 14:53.47 | DND | let me check |
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| 15:39.43 | scalex000 | leifmadsen, do you know another swift work with asterisk because the darren have errror when I compile |
| 15:51.54 | scalex000 | hello, I need help with this http://pastebin.com/khC00Cpn |
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| 16:17.49 | *** join/#asterisk moy_ (~moy@69.157.45.156) |
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| 16:57.29 | *** join/#asterisk irroot (~irroot@197.170.61.198) |
| 16:58.13 | irroot | evening |
| 16:58.18 | carrar | morning! |
| 16:58.28 | irroot | lol |
| 16:59.01 | irroot | is glad to be back in civilization .... did have good time in bush ... |
| 16:59.21 | Kobaz | i like bushes |
| 16:59.23 | WIMPy | prefers good time without bush. |
| 16:59.29 | Kobaz | shrubberies |
| 17:00.38 | irroot | was in http://pilanesberggamereserve.com/index.html ancient volcano now game reserve ... did not feed the kids to the lions was tempted ... |
| 17:03.01 | Kobaz | i was in west virginia doing some climbing over the weekend http://wildrockwv.com/blog/wp-content/uploads/2010/05/nrg-climbing.jpg |
| 17:03.28 | irroot | kobaz that is epic |
| 17:03.47 | irroot | nice view |
| 17:03.59 | Kobaz | that's not me, but that's where we were |
| 17:04.21 | carrar | haha |
| 17:04.24 | Kobaz | it's cool that the cliff starts high above the valley, so you just go up maybe like 50 feet and get a nice view |
| 17:05.50 | carrar | yeah I'm in seattle http://www.skydiveseattle.com/images/photo-gallery/expert-skydivers-in-formation.jpg |
| 17:06.01 | carrar | btw thats not me, but somewhere close too me |
| 17:06.06 | carrar | :) |
| 17:07.01 | Kobaz | heh |
| 17:07.14 | Kobaz | carrar: were you on the ground? |
| 17:07.25 | irroot | carrar apparently newton was not 100% right its not the fall that kills but the bounce ... need to grab the grass and hold on tight so not to bounce ... |
| 17:08.46 | scalex000 | i get this after i upgrade http://pastebin.com/KLvd1BvZ |
| 17:09.49 | Kobaz | http://a7.sphotos.ak.fbcdn.net/hphotos-ak-snc3/28203_399382108516_629843516_4413784_3500505_n.jpg |
| 17:09.53 | Kobaz | that's the actual me |
| 17:09.56 | Kobaz | nice butt-shot |
| 17:10.10 | carrar | Kobaz, somewhere in the same state :) |
| 17:10.12 | irroot | scalex000 need to bt the core |
| 17:10.23 | Kobaz | no, same formation as that guy in the picture |
| 17:10.32 | scalex000 | :S |
| 17:10.40 | scalex000 | bt? |
| 17:11.01 | Kobaz | I don't carry around a camera crew with me that can get the ariel shots |
| 17:11.03 | irroot | scalex000 gdb |
| 17:11.06 | *** join/#asterisk danboid (~dan@46-64-94-140.zone15.bethere.co.uk) |
| 17:12.56 | irroot | http://www.voip-info.org/wiki/view/Asterisk+debugging |
| 17:13.12 | danboid | What do people recommend for diagnosing SIP connection problems? |
| 17:13.20 | Kobaz | sip debug |
| 17:13.26 | Kobaz | and wireshark |
| 17:13.35 | *** join/#asterisk digiv_ (~mlhess@141.214.234.28) |
| 17:13.38 | irroot | https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information scalex000 |
| 17:14.18 | scalex000 | irroot, im unable to conect |
| 17:14.36 | irroot | there should be a core file ?? |
| 17:14.49 | danboid | Kobaz, OK - just looking into SIP debug now. More granny-proof and windows friendly the better here |
| 17:15.06 | Kobaz | sip debugging is not friendly |
| 17:16.20 | danboid | Does mumble work peer-to-peer or use a technique that means it 'just works' in most cases ala Skype allegedly does? |
| 17:17.12 | irroot | it would be mean to point danboid to the RFC ... |
| 17:17.21 | danboid | if not, what am I looking for in my quest for a skype and audio conferencing app that is non-network admin friendly |
| 17:17.44 | danboid | ie friendly to people who can't use wireshark and pals |
| 17:17.54 | danboid | does it exist yet? |
| 17:18.33 | irroot | skype have done a nasty recently and stoped asterisk support |
| 17:18.46 | irroot | there still alternatives |
| 17:20.03 | *** join/#asterisk g00gle (~thameema@c-98-248-232-219.hsd1.ca.comcast.net) |
| 17:29.07 | *** join/#asterisk sourcode (~code@ppp-58-8-220-245.revip2.asianet.co.th) |
| 17:31.16 | scalex000 | irroot im so dumb |
| 17:31.28 | irroot | dont admit it here :P |
| 17:32.54 | SunTsu | if you are people will find out anyway :> |
| 17:34.22 | scalex000 | lol |
| 17:34.30 | scalex000 | i can start my asterik |
| 17:34.40 | scalex000 | because safe_asterisk have error |
| 17:42.26 | *** join/#asterisk jkroon (~jkroon@dsl-242-11-09.telkomadsl.co.za) |
| 17:43.05 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 17:44.43 | *** join/#asterisk fofware (~fabian@host71.186-124-255.telecom.net.ar) |
| 17:45.25 | *** join/#asterisk mha_ (~mha@84.255.188.69) |
| 18:35.09 | *** join/#asterisk Lantizia (~Lantizia@erebus.seaquake.net) |
| 18:35.30 | *** join/#asterisk banditti (~banditti@unaffiliated/banditti) |
| 18:38.26 | *** join/#asterisk Lantizia (~Lantizia@erebus.seaquake.net) |
| 18:41.34 | *** join/#asterisk Lantizia (~Lantizia@erebus.seaquake.net) |
| 18:55.19 | *** join/#asterisk boazb (~b@85.250.144.255) |
| 18:59.14 | *** join/#asterisk mha_ (~mha@84.255.188.69) |
| 19:00.02 | mha_ | install asterisk in fedora, but cannot see it in KDE gui, any additional package for gui? |
| 19:01.51 | *** join/#asterisk JuStIcIa_ (~justicia@190.167.18.31) |
| 19:05.19 | WIMPy | What do you expect to see there? |
| 19:12.57 | mha_ | WIMPy: question to me? |
| 19:16.00 | ChannelZ | There is no GUI |
| 19:16.42 | ChannelZ | at least not in the traditional sense. You could get FreePBX to run on top of Asterisk but even that is a web-based thing and personally I think it's kind of a mess |
| 19:20.34 | jkroon | ChannelZ, kind of? |
| 19:20.46 | jkroon | you're being very polite. |
| 19:21.02 | *** join/#asterisk scalex000 (~chatzilla@186.6.134.200) |
| 19:22.42 | scalex000 | hello |
| 19:22.47 | scalex000 | i need help |
| 19:22.56 | scalex000 | my asterisk not start |
| 19:23.00 | scalex000 | I get error |
| 19:23.54 | SunTsu | scalex000: and you expect us to guess what error and how to fix it? ;) |
| 19:24.01 | ChannelZ | jkroon: Well I'm trying to improve my attitude :) |
| 19:24.07 | scalex000 | lol |
| 19:24.14 | scalex000 | not fix it |
| 19:24.26 | scalex000 | but I can find any help in forums |
| 19:24.28 | scalex000 | :P |
| 19:24.35 | SunTsu | scalex000: well, best start would be putting the error message onto pastebin and show us |
| 19:24.37 | scalex000 | I install again |
| 19:24.42 | scalex000 | but its the same |
| 19:24.45 | scalex000 | ies |
| 19:24.47 | scalex000 | yes |
| 19:24.55 | scalex000 | but is nobody answer why in eed to do |
| 19:25.05 | scalex000 | http://pastebin.com/KLvd1BvZ |
| 19:26.11 | SunTsu | scalex000: it's your problem, you need to give enough information so people can make themsalves a picture and decide that they want to help you and in fact are able to |
| 19:26.41 | SunTsu | If you don't put work into describing your error nobody will put work into helping you |
| 19:26.47 | scalex000 | did you see the paste bin |
| 19:27.10 | scalex000 | suntsu, I install the last version of asterisk |
| 19:27.24 | SunTsu | yeah, now. You might have gotten somebody to react if you had it done right from the start |
| 19:27.55 | scalex000 | suntsu, I have this version 1.6.0.20 working good |
| 19:28.01 | SunTsu | scalex000: well, segfault can be lots of different things. do you have gdb installed? |
| 19:28.15 | scalex000 | suntsu, I decide to install the last one because i can install the swift application |
| 19:28.30 | scalex000 | suntsu, yeah |
| 19:28.54 | scalex000 | suntsu, this automatic service start I not create by my self |
| 19:29.57 | SunTsu | scalex000: then look for the core. should be asterisk.core, and run gdb asterisk <path to core>/asterisk.core, enter "bt" in gdb and pastebin that |
| 19:30.27 | SunTsu | with you substituting <path to core> with the actual path, of course and "bt" without the wuotation marks |
| 19:30.43 | SunTsu | quotation even |
| 19:32.14 | scalex000 | :P |
| 19:32.33 | scalex000 | suntsu, Im noob |
| 19:32.47 | scalex000 | maybe I dont know where is the path |
| 19:32.49 | jkroon | ChannelZ, i hear you. but with at least two potential new clients per week asking me to support their "asterisk" (aka freepbx install) I tend to get rather ... uhrm, well, negative about it. so the official company policy is now "is it freepbx based?", "no, sorry, we will happily re-install ast for you from scratch but we won't take over your freepbx", "but it's asterisk", "no sir, sorry, we used to use freepbx but no longer" (imho it's |
| 19:32.50 | jkroon | <PROTECTED> |
| 19:32.58 | SunTsu | scalex000: if you don't understand something in specific ask about it |
| 19:33.30 | scalex000 | suntsu, core |
| 19:33.41 | scalex000 | suntsu, where i get the path |
| 19:34.01 | scalex000 | suntsu, its not asterisk -rx "show core" |
| 19:34.07 | SunTsu | scalex000: use find, like find / -name asterisk.core |
| 19:34.43 | jkroon | that could take a while ... |
| 19:35.11 | SunTsu | a core (dump) is what certain errors yield, it's a memory dump of the program at the point of the error |
| 19:35.56 | SunTsu | it's written to a file and gdb can use it to do a bt, a backtrace, which should show in what chain of function calls the program was |
| 19:36.23 | scalex000 | suntsu, not found nothing |
| 19:36.41 | jkroon | ooh crap. i just realized my one ast install is popping in chan_local.so (something called from ast_bridge_call) |
| 19:37.50 | SunTsu | scalex000: damn, the name depends, is asterisk a symbolic link to a differently named binary, like asterisk-1.6.andsoon |
| 19:38.44 | SunTsu | maybe the file is just calles core. you could do find / -name asterisk\*.core -o -name core |
| 19:38.50 | SunTsu | called even |
| 19:39.24 | SunTsu | that should find all files simply calles core or beginning with asterisk and ending in .core |
| 19:40.10 | scalex000 | suntsu, ok, i dont put .core etc |
| 19:40.56 | SunTsu | scalex000: er, sorry, what? |
| 19:41.05 | scalex000 | suntsu, hold on |
| 19:41.38 | SunTsu | unfortunately I must be going now, but somebody else sure can help |
| 19:41.50 | scalex000 | suntsu, i get a lot of asterisk |
| 19:41.55 | scalex000 | .pid |
| 19:42.15 | SunTsu | scalex000: that's why I wrote "asterisk\*.core" |
| 19:42.43 | scalex000 | suntsu, with core not found nothing |
| 19:42.51 | scalex000 | I remove .core |
| 19:45.42 | SunTsu | then your ulimit probably does not allow cores. do ulimit -c $((8096*1024)) and try again |
| 19:45.59 | SunTsu | run ulimit on itself and look what it says about corefiles |
| 19:46.16 | SunTsu | anyway, must be going, cu later |
| 19:51.40 | jkroon | anybody with ast development experience here? |
| 19:52.48 | jkroon | i've got a segfault on line 189 of chan_local.c, which references a write_info variable, and this segfaults, so either write_info is NULL, or the write_fn field is incorrectly populated. |
| 19:53.38 | scalex000 | unlimited |
| 19:53.53 | jkroon | ok, the fact that write_info is dereferenced earlier in the function it has to be the latter. |
| 19:54.01 | scalex000 | suntsu, unlimited |
| 20:06.42 | jkroon | hmm, could a gcc-bug be causing my issues? if I do a print write_chan->write_fn on the core it actually tells me cannot access memory at 0x8, however the frame indicates that the "data" parameter does have a value. which means that 0x8 is not even in the memory structure, when trying to print the raw value of write_info it tells me the value is optimized out ?!? |
| 20:12.17 | *** join/#asterisk Lantizia (~Lantizia@erebus.seaquake.net) |
| 20:13.12 | *** join/#asterisk olinux (~olinux@208.104.179.2) |
| 20:18.02 | *** join/#asterisk jc319 (~jc318@78-86-169-203.dsl.cnl.uk.net) |
| 20:18.41 | jkroon | or not ... chan_local is getting something passed to local_setoption that is def not a write_info structure. |
| 20:19.14 | jkroon | contains the raw string "IAX2", which probably indicates that there is some problem with bridging local and iax2 channels under some circumstances. |
| 20:23.33 | *** join/#asterisk zapotek6 (~zapotek6@83.224.73.28) |
| 20:35.34 | *** join/#asterisk Gugge (~gugge@91.208.16.1) |
| 20:35.45 | jc319 | silence... what's happening |
| 20:39.14 | scalex000 | in this version 1.6.2.18 how to reload all module |
| 20:39.17 | scalex000 | :P |
| 20:40.43 | jc319 | reload |
| 20:48.17 | olinux | any ideas why call connects but i have no audio |
| 20:48.33 | olinux | when dialing in i do hear main ivr greeting |
| 20:50.12 | *** join/#asterisk emsLinux (~dave@190.71.3.255) |
| 20:53.51 | emsLinux | Hello, there's someone who can help me with a Google Voice issue in my Asterisk Server, I think the problem is with the [googlein] context in extensions_custom.conf, are the configuration files related to the service and the problem explanaition http://bit.ly/kfE6FM |
| 20:58.50 | *** join/#asterisk Lantizia (~Lantizia@cpc17-stok16-2-0-cust23.1-4.cable.virginmedia.com) |
| 21:01.07 | *** join/#asterisk pdtpatrick___ (~pdtpatric@ip68-4-0-113.pv.oc.cox.net) |
| 21:09.49 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 21:18.27 | scalex000 | who have used swift for text to speech |
| 21:18.38 | scalex000 | I would like to know how to choose the voice I want |
| 21:20.13 | jkroon | olinux, dead phone? wrong routing? could be any of a few things. |
| 21:20.19 | jkroon | watch the CLI carefully. |
| 21:20.40 | jkroon | scalex000, swift vs festival ? |
| 21:21.05 | scalex000 | jkroon, no |
| 21:21.15 | jkroon | have you used both? |
| 21:21.21 | scalex000 | jkroon, nop |
| 21:21.52 | scalex000 | jkroon, I've installed swift |
| 21:21.54 | jkroon | kk, just wondering about performance and ease of use :) |
| 21:22.18 | scalex000 | jkroon, no, I want know how I can choose which voice in dialplan |
| 21:22.26 | olinux | thanks jkroon, i have switchvox system so interface is limitted, seems to affect all phones |
| 21:22.48 | scalex000 | jkroon, on voip.og show this syntax swift([voice],text) |
| 21:23.08 | jkroon | scalex000, there you go then :) |
| 21:23.13 | scalex000 | jkroon, but the app I've installed do not have |
| 21:23.26 | jkroon | olinux, tell it to not redirect audo to go direct and check again. |
| 21:24.39 | olinux | thanks jkroon will try to find that :) |
| 21:25.09 | jkroon | olinux, let's ask it this way - does internal calls work? |
| 21:25.13 | emsLinux | jkroon i don't think is something about routing, the CLI shows me nothing but "exited non-zero on 'Gtalk/+17607058888-9ff2'", everything seems to be ok, i dont know what to do, everything is working fine except incoming calls from Google Voice |
| 21:25.20 | emsLinux | yes, they are |
| 21:25.23 | olinux | jkroon, no |
| 21:25.37 | olinux | they connect but no audio on either end |
| 21:26.01 | jkroon | olinux, and switchvox doesn't give you access to the CLI either :p |
| 21:26.17 | olinux | it's actually here in version 5 |
| 21:26.34 | jkroon | emsLinux, up the verbosity and debug levels and check again? |
| 21:26.35 | olinux | but the "Start Debugging Session" dont seemt o work |
| 21:28.04 | scalex000 | jkroon, this syntax not exist |
| 21:28.24 | scalex000 | jkroon, I can only use default voice |
| 21:28.34 | jkroon | scalex000, do you have additional voices installed? |
| 21:28.46 | scalex000 | jkroon, yes |
| 21:29.08 | jkroon | can't help you then, switch(voicename,test here) should then work. |
| 21:29.45 | *** part/#asterisk emsLinux (~dave@190.71.3.255) |
| 21:32.32 | *** join/#asterisk moniker326 (~moniker32@c-71-56-135-99.hsd1.wa.comcast.net) |
| 21:34.26 | *** join/#asterisk networkuser (~S@78.129.237.130) |
| 21:34.54 | networkuser | in 1.8 I for blind transfer only I can press one digit !!?? |
| 21:35.18 | p3nguin | Can you please repeat that in English? |
| 21:36.55 | moniker326 | hello everyone. Installed asterisk/freepbx in debian and when I bring up the browser localhost/admin I just get the directories page no GUI |
| 21:36.58 | moniker326 | what am I missing? |
| 21:44.43 | DND | en moniker326 ty installing php |
| 21:44.49 | DND | *try |
| 21:45.35 | moniker326 | it says it's install |
| 21:45.37 | moniker326 | ed |
| 21:46.06 | DND | then in httpd.conf/ add the php module |
| 21:47.08 | DND | also this: AddType application/x-httpd-php .php |
| 21:47.08 | DND | AddType application/x-httpd-php-source .phps |
| 21:48.08 | DND | add also in DirectoryIndex .php |
| 21:48.41 | DND | then restart apache |
| 21:49.17 | *** join/#asterisk pdtpatrick___ (~pdtpatric@ip68-4-0-113.pv.oc.cox.net) |
| 21:49.26 | moniker326 | ok |
| 21:49.29 | moniker326 | just a min |
| 21:50.41 | moniker326 | I'm using apache2 |
| 21:50.45 | moniker326 | change it any? |
| 21:51.35 | moniker326 | or should it be httpd installed instead? |
| 21:52.01 | *** join/#asterisk Lantizia (~Lantizia@cpc17-stok16-2-0-cust23.1-4.cable.virginmedia.com) |
| 21:53.58 | moniker326 | in otherwords there's nothing in httpd.conf to add php module |
| 21:56.36 | *** join/#asterisk mmlj4 (~jkelly@ip24-252-121-95.no.no.cox.net) |
| 22:06.15 | *** join/#asterisk wonderworld (~ww@port-92-201-20-202.dynamic.qsc.de) |
| 22:26.24 | Wiretap7 | moniker326, I think DND is barking up the wrong tree |
| 22:26.36 | moniker326 | agreed |
| 22:26.50 | Wiretap7 | moniker326, however, if you're getting a directory listing, is index.php anywhere to be seen? |
| 22:27.08 | moniker326 | like where? |
| 22:27.16 | Wiretap7 | in the listing |
| 22:27.32 | moniker326 | listing of directories? |
| 22:27.37 | Wiretap7 | oh ffs |
| 22:27.45 | Wiretap7 | WHAT YOU SEE WHEN YOU LOOK AT YOUR SCREEN |
| 22:27.49 | Wiretap7 | IS INDEX.PHP THERE? |
| 22:27.58 | moniker326 | no |
| 22:28.04 | moniker326 | just index.html |
| 22:28.10 | Wiretap7 | click that |
| 22:28.17 | moniker326 | I did |
| 22:28.23 | moniker326 | just come up it works |
| 22:28.23 | Wiretap7 | and what is on it? |
| 22:28.29 | Wiretap7 | in that case |
| 22:28.36 | Wiretap7 | you have fucked up your apache config royally |
| 22:28.43 | Wiretap7 | go back to square one and start again :) |
| 22:28.44 | moniker326 | :( |
| 22:28.51 | moniker326 | k |
| 22:29.07 | moniker326 | used freedoh :( |
| 22:29.11 | Wiretap7 | lol |
| 22:29.13 | moniker326 | for debian |
| 22:29.24 | Wiretap7 | in that case... try localhost/freepbx/ I think it is |
| 22:29.29 | Wiretap7 | also, this isn't #freepbx |
| 22:29.42 | moniker326 | can't find it today...... |
| 22:29.57 | Wiretap7 | you're pretty hopeless at supplying the information people need to help you troubleshoot |
| 22:31.47 | moniker326 | yeah well I've tried a few tutorials online to try to install this on debian and each one is pretty lacking in information after initial script install |
| 22:31.57 | moniker326 | all act like it 'should work' as is |
| 22:32.05 | moniker326 | not talking about bugs |
| 22:32.30 | moniker326 | guess I'll go to the preinstall centos..... |
| 22:59.17 | *** join/#asterisk gruvfunk_afk (~chatzilla@cpe-68-172-221-157.hvc.res.rr.com) |
| 23:03.51 | jc319 | moniker326 just a suggestion - start with asterisk (no GUI) once you have the basics working, build on that (extended config or GUI whatever you wish). |
| 23:06.44 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 23:08.30 | Wiretap7 | jc319, that doesn't really work |
| 23:08.37 | Wiretap7 | you can't make a GUI work with your existing config |
| 23:10.34 | jc319 | Too bad for people who need GUI. What GUI are you using, if any? |
| 23:11.52 | Wiretap7 | I've been deploying FreePBX just because its easy to maintain |
| 23:12.13 | Wiretap7 | I apply a lot of customisations to support the Cisco phones I use |
| 23:12.31 | Wiretap7 | at some point I should probably start writing my asterisk configs from scratch |
| 23:14.02 | *** join/#asterisk Tech_Travis (~Travis@cpe-76-168-191-127.socal.res.rr.com) |
| 23:14.45 | jc319 | What kind of customizations do you use? |
| 23:21.29 | Wiretap7 | just a bunch of extra dialplan to support the features on the cisco phones |
| 23:22.50 | jc319 | I have Cisco phones but I think I can't use the extra features because I don't have SCCP, I just use SIP now - with a plan to use IAX at some point once I sort out my dial plan and fix my SIP issues |
| 23:23.12 | jc319 | Can I have an example config from you to see what's possible and what am I missing? |
| 23:27.47 | Haraken | jc319, hey man |
| 23:27.59 | jkroon | Wiretap7, come again? freepbx easy to maintain? wtf am I missing?!? |
| 23:28.01 | Haraken | jc319, I used a different distro and got my voicemail light to work :) |
| 23:29.03 | Haraken | any of you guys using cisco phones know any project that assists with setting up a services url? |
| 23:30.05 | jc319 | Haraken: Fantastic news |
| 23:30.36 | Haraken | it's nice having a voice mail light xD |
| 23:31.04 | Haraken | I've pretty much completed all of my initial goals for setting up a voip system in my house... now I am just going at it to see what else I can do |
| 23:32.09 | jc319 | This is one detailed page about XML features of 7960s http://www.ibm.com/developerworks/wireless/library/wi-voip/ |
| 23:33.53 | Haraken | thanks |
| 23:34.28 | Haraken | I found a neat project called asteristickies but it looks like it hasn't been updated in a couple years and the latest version doesn't seem to install properly... shame too because the youtube video I saw of it was pretty neat |
| 23:37.03 | jc319 | Is there a particular reason that you want to reflect some data into that tiny screen? There's some potential to develop cool stuff there but I can't concentrate on it due to the fact that I have 2 huge screens 10 inches away from that monochrome screen. If you have a good reason I might be interested |
| 23:38.08 | Haraken | I mainly just want to put a list of feature codes in there, that's about it |
| 23:38.34 | Haraken | but while searching for how to do that I found asteristickies which seemed like it would be fun to implement |
| 23:38.57 | Haraken | only other thing I want to do beyond that is make some ring tones :) |
| 23:39.12 | Haraken | or find some |
| 23:39.18 | Haraken | seems mine only has two ringtones at the moment xD |
| 23:42.25 | jc319 | http://www.loligo.com/asterisk/cisco/79xx/current/ |
| 23:49.02 | jc319 | http://home.earthlink.net/~jmkord/cisco_ring.htm and you can make your own tunes using this http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/3_0_9/a3rings.html |
| 23:50.44 | *** join/#asterisk sarthor (~sarthor@unaffiliated/sarthor) |
| 23:53.33 | sarthor | Hi, I did install asterisk on my ubuntu-server 11.04 with apt-get install asterisk, and installed successfully, now i can not browse from client pc, as i am doing http://192.168.30.1:8088, i defined same ip and port in /etc/asterisk/http.conf file, Any help please, |
| 23:55.59 | ChannelZ | asterisk has no GUI |
| 23:56.27 | *** join/#asterisk superbofh (mouse@2001:b18:4059:0:8c7d:9a6b:e322:8c79) |
| 23:56.53 | sarthor | ChannelZ, but can we not browse asterisk as a webpage? |
| 23:57.05 | ChannelZ | not by default |
| 23:57.30 | ChannelZ | There are 3rd party things like FreePBX if you really want such things |
| 23:58.50 | sarthor | Hmm. i actually want to learn how to configure/run asterisk server, and i have ubuntu server 11.04 installed, So i am compell to install on that machine. |
| 23:59.56 | sarthor | ChannelZ, I removed that asterisk, that was 1.6 version, Can you please look here, http://letitknow.wordpress.com/2011/05/05/how-to-install-asterisk-1-8-on-ubuntu-server-11-04/ , is it good to follow? |
