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| 02:47.06 | xxiao | hmm 2.6.18 |
| 02:47.11 | xxiao | long time no see |
| 02:57.31 | xxiao | retrieve_conf FATAL?? |
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| 04:57.58 | sawgood | Fresh new install of 1.8.4 from RPM on a new CentOS box, and I am getting this error from the CLI |
| 04:58.03 | sawgood | [Jun 3 21:48:13] WARNING[2927]: db.c:112 dbinit: Unable to open Asterisk database '/var/lib/asterisk/astdb': Permission denied |
| 04:58.06 | sawgood | any tips? |
| 04:58.18 | sawgood | the file permissions for this are exactly like another box I have |
| 05:00.51 | sawgood | asterisk is the owner of /var/lib/asterisk |
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| 05:11.27 | sawgood | no suggestions, huh? |
| 05:16.01 | sawgood | got it! |
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| 05:17.24 | Valen | what was it? |
| 05:17.26 | p3nguin | I guess the permissions weren't "exactly" like the other box, eh? |
| 05:18.33 | Valen | p3nguin: so yah still having difficulty getting anything to run after the "hangup" call |
| 05:19.56 | sawgood | no they were not |
| 05:20.07 | sawgood | just off a hair! |
| 05:20.32 | Valen | figured they would be off by a ...bit |
| 05:20.33 | p3nguin | I'd guess there was a problem with /var/lib/ or /var/lib/asterisk/ having the wrong owner. |
| 05:20.35 | Valen | runs |
| 05:20.50 | sawgood | p3nguin: how'd ya know? |
| 05:21.24 | p3nguin | It seems to be common that /var/lib/asterisk/ often doesn't have the right owner for asterisk running as user asterisk. |
| 05:21.45 | Valen | exten => *12,n(hook_off),Hangup |
| 05:21.45 | Valen | exten => h,1,Verbose(call just hung up) |
| 05:22.02 | Valen | any idea why the "just hung up" part wouldn't get run? |
| 05:22.18 | p3nguin | I don't mean common for the rpms, since I don't use the rpms, but it seems like a common thing to encounter. |
| 05:22.26 | sawgood | I now have Digium G.729 codec licenses running smoothly ... |
| 05:22.49 | Valen | sawgood: nice, worth it for a small office do you think? |
| 05:23.20 | sawgood | well for ADSL with only 384kb up .... it was either G.729 or an IAX2 trunk |
| 05:23.28 | p3nguin | valen: Is your call starting in the context where those two extensions are? |
| 05:23.59 | Valen | p3nguin: i just barely comprehend the question |
| 05:24.10 | Valen | I'm dialing *12 |
| 05:24.31 | p3nguin | Within what context are those extensions? |
| 05:24.56 | Valen | [app-userlogonoff] |
| 05:25.04 | Valen | http://pastebin.ca/2074356 you can see the whole thing there |
| 05:26.37 | p3nguin | Take a look at line 22. |
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| 05:27.36 | Valen | yeah the macro? |
| 05:27.48 | p3nguin | h is being run. |
| 05:27.59 | p3nguin | But it's h from the "parent" context rather than h in the final context. |
| 05:28.47 | Valen | that wooshing noise was that going over my head lol |
| 05:29.17 | p3nguin | Your phone has to be assigned to a context where all calls from the phone will begin. |
| 05:29.30 | p3nguin | It appears that your phone is in the from-internal context. |
| 05:29.39 | Valen | that would make sense yes |
| 05:30.32 | p3nguin | It seems like *12 is being matched in another context in another file rather than this one you've shown me. |
| 05:31.13 | Valen | if i put something like *12,2,Verbose(stuff) |
| 05:31.20 | Valen | then i get something happening |
| 05:31.37 | Valen | IE i get "stuff" in the cli |
| 05:32.13 | Valen | perhaps i'm better off inserting into the main swing of things before the hangup stuff happens |
| 05:32.33 | p3nguin | When you dial *12, the call starts out in from-internal and has to either match an explicit extension in from-internal, a pattern extension in from-internal, or one of those types of extensions within a context which has been included in from-internal. |
| 05:34.21 | p3nguin | In the output you showed me, it is matching somewhere else besides the context you created. That usually happens because there is a match before your file is included. There is a specific order that extensions are "read into" pbx_config. |
| 05:34.50 | Valen | greping /etc/asterisk for *12 only shows those entries |
| 05:35.02 | p3nguin | The reason your file is not being used is directly related to the fact that you are using FreePBX. |
| 05:35.11 | Valen | well the file is being used |
| 05:35.30 | Valen | if i put this in exten => *12,2,Verbose(call just ${CALLERID(num)}) it works |
| 05:35.58 | p3nguin | Show me the context containing your #Include for the file which contains your new context. |
| 05:36.00 | Valen | ie i can see that in the cli |
| 05:36.15 | p3nguin | hmm |
| 05:36.23 | p3nguin | Show me that part working as you've described. |
| 05:36.42 | p3nguin | Wait, nevermind. |
| 05:36.49 | p3nguin | You already showed me that part working. |
| 05:36.55 | p3nguin | # -- Executing [*12@from-internal:2] Hangup("SIP/9999-000000b4", "") in new stack |
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| 05:37.17 | p3nguin | *12, priority 2. Hangup() |
| 05:37.28 | Valen | http://pastebin.ca/2074460 |
| 05:38.04 | Valen | all i really want to do is send a sip notify after the call is over |
| 05:38.29 | p3nguin | Okay, this confirms that the reason it isn't doing what you want is because h is being run in the starting context. |
| 05:39.05 | p3nguin | If you modify the h extension within from-internal, you'll get different results. |
| 05:39.36 | Valen | that sounds like a bad idea to my mind |
| 05:39.58 | p3nguin | I wouldn't modify it there. |
| 05:40.00 | Valen | i think i'll try and insert before the hangup call with a fixed delay and go from there |
| 05:40.50 | p3nguin | I would, however, see if you can use a Goto() to get you into another context and see if it still executes the h extension from the starting context. |
| 05:41.07 | Valen | I'd rather not mess with the actual flow too much |
| 05:41.16 | Valen | it probably does something important |
| 05:41.37 | Valen | is there a convenient method of doing that within the dialplan (ie run this command after a delay of x, but don't halt execution of the rest of it?) |
| 05:41.45 | p3nguin | Yep, that's what FreePBX does... takes over everything so you can't modify what you want when you want. |
| 05:42.23 | Valen | it also means somebody like me can get it all working and i only run into trouble when i try and do wild and crazy stuff, so its a net win for me ;-> |
| 05:44.20 | Valen | I'd better get going to take the dogs for a walk, i'll bbs |
| 05:45.11 | Valen | if i don't see you again p3nguin i really appreciate your help so far, it would have taken me ages to even get to here |
| 05:45.50 | p3nguin | I do what I can. Someone will surely come along and get you where you need to go. |
| 05:54.11 | p3nguin | I put a couple images in your directory that you can use. |
| 05:54.19 | p3nguin | Err... wrong window. |
| 06:19.52 | ChannelZ | I hope it's not of your weiner. |
| 06:20.31 | p3nguin | :( |
| 06:21.08 | ChannelZ | "I'm probably almost sort of sure that picture isn't of me." |
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| 09:31.21 | irroot | anyone here having problem with missing DTMF digits ?? |
| 09:33.03 | WIMPy | Maybe you should tell us, where they go missing. |
| 09:37.39 | irroot | to the great big bit bucket in the sky :P |
| 09:37.55 | irroot | M19060 |
| 09:38.04 | irroot | https://issues.asterisk.org/view.php?id=19060 |
| 09:39.00 | WIMPy | So it's SIP. |
| 09:39.31 | irroot | not exclusivly |
| 09:39.58 | irroot | the problem is that when sending DTMF fast it doe not transmit all the digiits |
| 09:40.26 | irroot | it could affect all techs |
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| 11:05.36 | ectospasm | woohoo! Four years working for Digium! |
| 11:05.43 | ectospasm | ...as of TODAY! |
| 11:08.22 | sxpert | ectospasm: guess it's time to go, fetch cake & champagne for your nice co-workers ;) |
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| 11:12.38 | ectospasm | unfortunately they don't do alcohol at work |
| 11:14.34 | sxpert | hah... americans ... |
| 11:14.48 | sxpert | no alcohol, no women, no nothing ^^ |
| 11:16.07 | irroot | that is bad had couple beers and a BBQ [Braai] yesterday afternoon do so regularly cant think of work with out a chill down |
| 11:16.21 | *** join/#asterisk KNERD (~KNERD@99.61.126.92) |
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| 11:18.11 | ectospasm | well, we'll get together outside of work and drink. Unfortunately the grave litigiousness of the US, with its associated insurance boom, practically disallows that kind of corporate sponsored party. That or our corporate culture, unfortunately |
| 11:18.24 | KNERD | What's this all about when trying to run, MAKE MENUSELECT---> Terminal must be at least 80 x 27 |
| 11:19.00 | ectospasm | the menu will look funny otherwise, so they force it to be that size |
| 11:19.17 | ectospasm | if you can't stretch your terminal window to be at least that size, I dunno |
| 11:19.20 | KNERD | so why does it not do it then? |
| 11:19.32 | KNERD | yeah it did |
| 11:19.40 | *** part/#asterisk jhmichelle352 (~jhmichell@41.206.146.255) |
| 11:19.45 | KNERD | i stretched it even a lot more |
| 11:19.57 | WIMPy | Maybe too far? |
| 11:20.04 | ectospasm | KNERD: you may need to restart it, it might read the terminal geometry only once. |
| 11:20.06 | KNERD | it functions now |
| 11:20.54 | ectospasm | 80x27 is the virtual terminal geometry in Linux, well standard that I've seen anyway. Or maybe it's greater than 27 lines, can't remember, but it sounds about right. |
| 11:21.08 | KNERD | first time I have ever saw that |
| 11:21.33 | WIMPy | I think it's 25 or 32 lines. |
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| 11:24.40 | KNERD | thanks |
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| 13:20.49 | leifmadsen | ectospasm: I had no idea you worked at digium :) |
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| 13:43.49 | xxiao | is mcumediaserver project the same as asterisk-video? |
| 13:44.33 | xxiao | test |
| 13:44.46 | *** part/#asterisk xxiao (~xxiao@li41-126.members.linode.com) |
| 13:44.49 | *** join/#asterisk xxiao (~xxiao@li41-126.members.linode.com) |
| 13:44.53 | xxiao | test |
| 13:45.25 | xxiao | is mcumediaserver the same thing as asterisk-video? |
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| 14:05.22 | danboid | I casn't see where I enter my SIP server details into linphone- its just stuck @192.168.1.65 or something? |
| 14:08.03 | danboid | When I go on 'configure a SIP account' it doesn't like any value I give for my SIP identity |
| 14:08.22 | danboid | what should I enter here? |
| 14:10.16 | danboid | maybe thats the incorrect place to enter my account details. If there is a better cross platform SIP client then I'd like tpo know about it but linphone looks ideal to me so far (apart from this :) |
| 14:24.59 | *** join/#asterisk jayvee (~jayvee@azaroth.sunriseroad.net) |
| 14:26.21 | jayvee | so...it turns out there is exactly zero documentation on the asterisk jingle channel, which is exactly what I'm trying to get going |
| 14:26.55 | jayvee | getting "WARNING[3218]: chan_jingle.c:1523 jingle_request: Could not find recipient." when trying to originate a channel |
| 14:27.28 | jayvee | I'm guessing I am giving it the wrong name of the person to call, but I have no idea how to find out what names it *should* have |
| 14:27.42 | jayvee | > channel originate Jingle/asterisk@sunriseroad.net/jeremy@sunriseroad.net extension 2324 |
| 14:27.48 | jayvee | > channel originate Jingle/asterisk/jeremy@sunriseroad.net extension 2324 |
| 14:27.51 | jayvee | both crap out |
| 14:28.09 | jayvee | if I just use "Jingle/jeremy@sunriseroad.net", it complains that there are "Bad arguments in Jingle Dialstring" |
| 14:28.51 | jayvee | been reading http://svn.asterisk.org/svn/asterisk/branches/1.8/channels/chan_jingle.c |
| 14:28.54 | jayvee | static struct jingle_container jingle_list; |
| 14:29.18 | jayvee | I'd like to know what is inside jingle_list, but there is no "jingle show recipients" sort of command |
| 14:32.03 | *** part/#asterisk cedwards (~cedwards@unaffiliated/zelut) |
| 14:33.33 | jayvee | jingle = ASTOBJ_CONTAINER_FIND(&jingle_list, name); |
| 14:33.48 | jayvee | name is set to the bit after the / |
| 14:34.01 | jayvee | so in this case, jingle = ASTOBJ_CONTAINER_FIND(&jingle_list, "jeremy@sunriseroad.net"); |
| 14:34.31 | jayvee | I'm not sure what an "ASTOBJ_CONTAINER" is, nor do I know how to enumerate it manually |
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| 14:54.48 | xxiao | danboid: try ekiga |
| 14:54.59 | danboid | xxio |
| 14:55.06 | xxiao | linphone just crashed on my 10.04 ubuntu 64bit, could not even get it running after installing |
| 14:55.25 | jayvee | odd, I use linphone on 10.04 all the time |
| 14:55.25 | danboid | xxiao, I have - no good. ekiga doesn't play nice with pulseaudio |
| 14:55.41 | danboid | I got my linphone prob sorted in the end |
| 14:55.53 | xxiao | what is it, which sip account are you using |
| 14:56.07 | xxiao | i could not get sip2sip.info on empathy either |
| 14:56.08 | danboid | but not got to try it yet! Can I call one of you to quickly test it please? |
| 14:56.57 | xxiao | i think i only have earphone without a mic at this point |
| 14:57.23 | xxiao | do they have test number like what skype has? |
| 14:57.38 | xxiao | you can call that number to check the setups |
| 14:57.49 | danboid | xxiao, Yeah- I've tried an echo test and that went ok but I want to try it for real now |
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| 14:58.32 | xxiao | danboid: still learning the whole voip here, guess I need my router to forward the port first, never made any sip calls yet |
| 14:59.11 | danboid | OK, no-one else willing to take a 30s (if that) call in all of asterisk land today? |
| 14:59.50 | xxiao | i'm willing to, until i set it up :) |
| 15:00.08 | danboid | xxiao, Thanks for your enthusiasm! :) |
| 15:00.10 | xxiao | is adding sip2sip.info account to linphone now |
| 15:00.35 | xxiao | it appears after i unplug my webcamera linphone stays alive |
| 15:00.42 | xxiao | otherwise it could crash again |
| 15:02.08 | danboid | xxiao, so its a no-go? I'm not bothered about video- just want to make a real voice call |
| 15:02.22 | xxiao | added account to linphone just now |
| 15:02.35 | xxiao | trying to set up NAT forwarding at this point |
| 15:04.10 | xxiao | you can try to call xxiao@sip2sip.info now |
| 15:04.21 | xxiao | i can at least hear if it goes through |
| 15:05.17 | xxiao | i heard you! |
| 15:05.21 | xxiao | sorry can't speak |
| 15:05.31 | danboid | Thanks! |
| 15:05.35 | xxiao | cool |
| 15:06.12 | danboid | xxiao, Well its proved something but I still need to make a 2-way call before I'm happy! Thanks anyway! |
| 15:06.34 | danboid | xxiao, I need to know how good/bad the latency is |
| 15:06.35 | xxiao | my first sip call here, well , half of it, thanks |
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| 15:11.53 | danboid | OK, looks like I've got a mate willing to try linphone with me now.. |
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| 15:20.29 | xxiao | danboid: have you seen this: You can try if you are reachable from the world dialing this number: sip:520@ekiga.net You will shorly be called by sip:500@ekiga.net after dialing this Call Back test number. It support audio and video. |
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| 17:01.39 | ejcweb | Hi - I'm not sure if this is the right channel for this, but... I'm thinking about starting to use VOIP/SIP - are there free services that provide you with accounts? I'm not entirely clear how it works. Thanks! |
| 17:04.00 | Kobaz | freeworlddialup |
| 17:04.57 | Kobaz | or, hmm |
| 17:05.00 | Kobaz | i guess they are defunct |
| 17:05.07 | Kobaz | it wasn't free to land lines though |
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| 19:21.20 | Kobaz | http://www.marlaolmstead.com/ |
| 19:21.44 | Kobaz | whenever i have kids, they are going to start abstract painting at 4 |
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| 19:29.57 | stram | are there any DiD providers which don't keep dialed number logs, incoming and outgoing? i'm getting sick of knowing that sipgate has a log of every call i make/receive. |
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| 19:36.06 | rhollan | Q: Where do asterisk sound files go for version 1.6? |
| 19:36.15 | rhollan | /var/lib/asterisk/sounds does not seem to work |
| 19:37.40 | *** join/#asterisk gavimobile (~user@82.166.237.204) |
| 19:38.54 | gavimobile | what os should do you guys recommend for my asterisk server. I would like to use it for my busniess and I will be building this server. I want to use sip |
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| 20:10.10 | xxiao | gavimobile: it depends on which platform you like, both redhat/centos and debian/ubuntu seem fine |
| 20:10.46 | xxiao | i'm building one on ubuntu lucid now, actually it's pre-packaged but i need make sure the configs work for me, so far so good |
| 20:11.19 | xxiao | for a mid size business asterisk should work well |
| 20:17.20 | gavimobile | I don't have a preference |
| 20:18.04 | gavimobile | I was more thinking of debian though |
| 20:18.23 | gavimobile | although ubuntu community has a lot of support |
| 20:18.35 | gavimobile | who has the most updates |
| 20:18.37 | gavimobile | ? |
| 20:25.43 | xxiao | if asterisk in debian stable? |
| 20:26.03 | xxiao | s/if/is |
| 20:26.26 | gavimobile | someone from freepbx recommended debian 6, its what he uses and he's one of the gurus of the channel |
| 20:26.28 | xxiao | anyway i prefer ubuntu/debian but centos is fine as well |
| 20:26.51 | xxiao | well i hope he had a blog on any debian6 tweaks, if any |
| 20:27.08 | xxiao | don't have a debian6 nearby though,can give it a try in vbox |
| 20:27.11 | gavimobile | but I guess the most important thing is support |
| 20:27.21 | gavimobile | which distro has the most support |
| 20:27.25 | xxiao | well then do ubuntu LTS |
| 20:27.43 | xxiao | maybe just hook up with asterisk guys |
| 20:27.49 | gavimobile | what does the lts staned for |
| 20:27.55 | xxiao | long-term-support |
| 20:28.09 | xxiao | ubuntu releases those better-tested release for 5 year support |
| 20:28.34 | xxiao | ubuntu 10.04 LTS is the one i'm using |
| 20:28.35 | gavimobile | can you give me a link, I think I want a minimal version |
| 20:28.46 | xxiao | google is your friend |
| 20:28.47 | gavimobile | a server edition without a gui |
| 20:29.00 | xxiao | you can do that certainly with this LTS |
| 20:29.16 | gavimobile | xxiao: ok. yea I can use google but I don't want to download a wrong version |
| 20:29.25 | gavimobile | ok thanks a lot |
| 20:29.46 | xxiao | http://releases.ubuntu.com/lucid/ |
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| 20:41.17 | gavimobile | xxiao: this is what I found with google http://releases.ubuntu.com/8.04/ubuntu-8.04.4-server-i386.iso |
| 20:41.43 | gavimobile | should I get this one instead http://releases.ubuntu.com/lucid/ubuntu-10.04.2-server-i386.iso |
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| 20:57.00 | ujjain | Hi, what is good software that supports pre- and post-paid billing? :) |
| 20:58.18 | ujjain | any reason not to take a2billing? |
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| 21:08.24 | xxiao | gavimobile: yes use the 10.04 one which was release 2010.04 |
| 21:08.53 | xxiao | that has a way better Asterisk pre-packaged |
| 21:09.10 | xxiao | is going to take a nap |
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| 21:10.06 | Kage` | Got a fun one for you guys: |
| 21:10.35 | Kage` | We have Syncterm (Synchronet BBS) installed, and Asterisk, on the same FreeBSD Jail. We want to run the BBS through a DID. Thoughts? |
| 21:11.18 | Kage` | My simplest and most brute-forced idea is a 127.0.0.1 telnet into syncterm, and just transmit that via modem signals over the DID, but does such a possibility exist for Asterisk? |
| 21:11.18 | Nugget | telnet is eeeeeeevil! |
| 21:17.26 | showme | I've looked around for a "sip modem" and have yet to find one, I've seen lots of "sip fax modems", but no sip modems.. It seems like it wouldn't be that hard to write |
| 21:20.43 | Kage` | Why has no one written one yet? |
| 21:20.49 | Kage` | I would have no idea where to begin, though |
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| 21:36.26 | Gugge | maybe because the SIP voice stream is compressed to much for a usable modem signal |
| 21:38.06 | showme | I can get a real modem to work over u-law with reasonable success |
| 21:39.09 | showme | I think the reason no one has written one is because most people think if you already have your internet connection, you don't need a modem, just do it over the internet.. They miss the point that not everything is on the internet |
| 21:41.51 | Nugget | if something isn't on the internet it isn't real. |
| 21:44.59 | showme | all of my core routers are in the internet, but if I can't ping them I sure need another way to hit them... |
| 21:48.20 | Gugge | if i cant ping my routers, i just disable the firewall :P |
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| 23:14.14 | showme | Kage: to answer your question, once someone develops a "sip modem", your request should be pretty trivial. |
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| 23:56.34 | Kage` | showme, Who could I ask to work on such a thing? I'm grossly underqualified to do so |
| 23:57.08 | Kage` | I know the whole, "Email the list" shot in the dark approach, but I'm wondering if I could approach somenoe specifically |