| 00:04.06 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 00:04.44 | lorenzo | any known issues of asterisk 11 not resolving SRV records? |
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| 03:23.42 | mnathani | what do I need to put in the spool directory to get asterisk to make a call from the command line? |
| 03:31.41 | ChannelZ | google 'asterisk call files' |
| 03:32.21 | ChannelZ | or you can use 'channel originate' perhaps on the console, or use asterisk -x to run that from a CLI (script or something) |
| 03:32.34 | ChannelZ | Or you could write something in AMI |
| 03:32.41 | ChannelZ | Lots of ways to stone that bird |
| 03:46.17 | *** join/#asterisk colocate (~colocate@unaffiliated/colocate) |
| 03:50.39 | colocate | Hoping for suggestions, I have setup a new SIP provider on a 1.6.2.9-2+squeeze6 box. I see connections coming from both the previously existing provider and the new provider. Previous provider is still working fine. With the new provider I can dial the DID that routes into the server. The call will route to a soft phone. The soft phone answers but the far end call never sees the call as being answered. Any thoughts on where the breakdown i |
| 03:57.04 | stevePearPear | hi im using realtime for sippeers, is there any chance for me to reload the data without reloading the module chan_sip? |
| 03:57.36 | stevePearPear | when i updated the database, with new sip credential, if I donât reload chan_sip, the data doesnât seems to be updated |
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| 04:42.09 | dsr | hi, is there a way to get logged in user's main channel in any context in the dialplan? |
| 04:42.56 | dsr | in the cli i can run agent show 123 and the channel is shown in the output as LoggedInChannel |
| 04:43.06 | dsr | I wanted to get this value in the dialplan |
| 04:43.10 | dsr | is there any way to do it? |
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| 06:17.15 | mnathani | ChannelZ: thanks, I got the call files working |
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| 06:26.07 | opsview | does anyone have experience with asterisk E1 trunk? |
| 06:26.49 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 06:26.56 | v0lZy | Hello |
| 06:27.27 | v0lZy | I'm having a problem I don't know how to deal with ... incoming calls from the outside world drop after about 3 min |
| 06:27.41 | v0lZy | I set pendantic=no in [general] sip.conf ... no change. |
| 06:28.35 | v0lZy | erm.. sorry, durration of last call was 01:36 |
| 06:31.03 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 06:32.55 | v0lZy | Second call dropped at 01:37 .. |
| 06:33.01 | v0lZy | Any ideas what would be causing this? |
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| 06:42.35 | v0lZy | hm, this seems to happen every 90 seconds or so |
| 06:42.48 | v0lZy | and thats the default session-minse |
| 06:43.29 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:11b3:646a:e7fc:964c) |
| 06:48.28 | v0lZy | if i set session-timers=refuse ... what negative consequences am i looking at? |
| 06:51.05 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 06:51.12 | spicyramen_ | If I remember correctly wont be a SIP refresh (re-invite) |
| 06:51.31 | v0lZy | and what does that do? |
| 06:51.35 | spicyramen_ | the negative effects would be calls not dropping in case there are failures |
| 06:51.47 | v0lZy | ah, so just silence |
| 06:51.57 | spicyramen_ | lets say A calls B, B drops but for whatever reason no BYE is sent |
| 06:52.01 | *** join/#asterisk opsview (edwinlee@gateway/shell/devio.us/x-otqedjqivndgvqms) |
| 06:52.10 | spicyramen_ | as * still think there's an active call |
| 06:52.12 | v0lZy | i dont know why im having this 90 seconds issue, but i set session-timers=refuse on the trunk, and now it seems to not drop |
| 06:52.12 | opsview | Hello! |
| 06:52.14 | spicyramen_ | u may see just phantom calls |
| 06:52.30 | v0lZy | wonder if there's other values that fix this without the BYE issue.. |
| 06:52.31 | spicyramen_ | yes, I would leave it on the trunk only |
| 06:52.37 | spicyramen_ | I would test Hold/resume |
| 06:53.12 | spicyramen_ | when the call drops was there any SIP message ? |
| 06:53.17 | spicyramen_ | before u made the change |
| 06:53.47 | v0lZy | i didnt have debug on |
| 06:53.57 | v0lZy | but it just said exited non zero on the dial command.. |
| 06:55.24 | v0lZy | if i have session-timers=accept |
| 06:55.31 | v0lZy | or session-timers=originate |
| 06:55.46 | v0lZy | could i be experiencing drops because of incorrect session-minse value? |
| 06:56.06 | v0lZy | because it seems to drop every 90 seconds if thats on... or roughly at 90 anyway |
| 06:56.14 | spicyramen_ | that could be I would look into https://www.rfc-editor.org/rfc/rfc4028.txt |
| 06:56.20 | v0lZy | I'm thinking, if I have session-timers originate and the other end has refuse.... |
| 06:56.26 | spicyramen_ | enable the debug sip |
| 06:56.34 | v0lZy | then my calls probably drop if i dont get a message? |
| 06:57.24 | spicyramen_ | is that for calls originated from asterks ? |
| 06:57.26 | spicyramen_ | asterisk |
| 06:59.38 | v0lZy | no |
| 06:59.53 | v0lZy | When I call out, everythings fine |
| 07:00.13 | v0lZy | incoming calls from people beyond my asterisk are the ones that get dropped after said 90 seonds. |
| 07:00.16 | v0lZy | seconds? |
| 07:00.21 | v0lZy | seconds* darn keyboard.. |
| 07:00.38 | spicyramen_ | I see, so Asterisk is the UAS for the incoming call |
| 07:01.07 | spicyramen_ | the session timers are observed when call is established first |
| 07:01.29 | v0lZy | UAS .. user agent ... S? |
| 07:01.29 | spicyramen_ | can u try to enable SIP debug and use pastebin |
| 07:01.36 | spicyramen_ | server |
| 07:02.01 | spicyramen_ | I have seen this error before, just want to see exactly whats happening to give a more educated answer |
| 07:02.14 | v0lZy | ok, let me see if i can manage |
| 07:02.17 | spicyramen_ | k |
| 07:03.22 | v0lZy | i just ran rasterisk -vvvvdddd hope thats ok |
| 07:03.46 | v0lZy | hm |
| 07:03.52 | v0lZy | doesnt seam that shows sip.. hold on |
| 07:05.44 | spicyramen_ | asterisk -r |
| 07:05.46 | spicyramen_ | then sip set debug on |
| 07:06.01 | v0lZy | yeah, got it :D |
| 07:06.10 | v0lZy | lots of sip traffic there, humph |
| 07:06.36 | spicyramen_ | yeah off it |
| 07:06.39 | spicyramen_ | on it place call |
| 07:06.44 | spicyramen_ | off it |
| 07:07.27 | v0lZy | off it when? |
| 07:07.34 | v0lZy | when it terminates? |
| 07:07.36 | v0lZy | after 90 seconds |
| 07:07.39 | v0lZy | thats gonna be a lot of sip .d |
| 07:08.18 | v0lZy | ok, it just ended |
| 07:08.23 | v0lZy | lets see if i can find anything |
| 07:08.27 | v0lZy | what am i looking for? |
| 07:08.36 | spicyramen_ | <PROTECTED> |
| 07:08.42 | spicyramen_ | and use the ip of the remote trunk |
| 07:08.52 | spicyramen_ | we are looking for Session timers |
| 07:09.00 | spicyramen_ | when call starts it is decided |
| 07:09.14 | spicyramen_ | who will be sending the session refresh |
| 07:09.27 | spicyramen_ | based on the min-Se, role, session timers |
| 07:09.43 | spicyramen_ | so first part of the call will help |
| 07:09.51 | spicyramen_ | and then after 90 secs we may see the error |
| 07:11.00 | v0lZy | i think i found something |
| 07:11.28 | stevePearPear | Hi, I am connecting my Asterisk to a sip trunk. is there anyway I could verify the sip trunk credential that my user has given me with the sip trunk? |
| 07:11.33 | v0lZy | i found the ending (but not the start, will do a different capture for that) |
| 07:12.48 | v0lZy | cheduling destruction of SIP dialog '06d26b495a2bd1c6117e4aee3eabefad@192.168.1.3:5060' in 32000 ms (Method: INVITE) |
| 07:14.01 | v0lZy | before that, read from UDP blablabla request timed out |
| 07:14.08 | *** join/#asterisk mirela666 (~mirko.bra@95.180.116.173) |
| 07:14.36 | spicyramen_ | ok yesâ¦looks like the negotiation at the begiinning is not succesful |
| 07:15.37 | v0lZy | so they dont negotiate the timers correctly |
| 07:15.51 | v0lZy | is that something the provider has to take care of, the originator, or me? |
| 07:15.55 | spicyramen_ | looks like, lets get those messages |
| 07:16.02 | spicyramen_ | it could be either |
| 07:16.10 | v0lZy | messages form start of call? |
| 07:16.17 | spicyramen_ | I would assume asterisk is waiting on the provider |
| 07:16.23 | spicyramen_ | and provider is waiting for u |
| 07:16.31 | spicyramen_ | and no refresh happens |
| 07:16.34 | spicyramen_ | hence fails |
| 07:16.37 | spicyramen_ | just a shot in the dark |
| 07:16.49 | spicyramen_ | I used to see this a lot with our app and Cisco TelePresence MCUs |
| 07:16.57 | spicyramen_ | they wanted always to send the session refresh |
| 07:17.36 | spicyramen_ | yes v)lzy |
| 07:18.53 | spicyramen_ | which * version are you using ? |
| 07:19.40 | spicyramen_ | in initial invite from your SP |
| 07:19.45 | spicyramen_ | you should see something like this: |
| 07:19.46 | spicyramen_ | Session-Expires: 90;refresher=uac. |
| 07:19.46 | spicyramen_ | Min-SE: 90. |
| 07:19.46 | spicyramen_ | Supported: timer. |
| 07:20.07 | spicyramen_ | and then u want to see 200 Ok from asterisk |
| 07:21.03 | v0lZy | ok, i think i found something |
| 07:21.33 | spicyramen_ | k |
| 07:21.57 | v0lZy | Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO |
| 07:21.57 | v0lZy | Supported: replaces, timer |
| 07:23.02 | spicyramen_ | I need the whole SIP invite |
| 07:23.09 | spicyramen_ | and 200 OK |
| 07:23.16 | spicyramen_ | not just a line |
| 07:23.23 | v0lZy | yes, im still scrolling |
| 07:23.26 | spicyramen_ | k |
| 07:23.28 | spicyramen_ | use pastebin |
| 07:23.49 | v0lZy | found it |
| 07:23.50 | v0lZy | Session-Expires: 180;refresher=uas |
| 07:24.54 | spicyramen_ | is that the initial invite? |
| 07:25.11 | spicyramen_ | from your SIP SP ? |
| 07:27.00 | *** join/#asterisk r00f (~r00f@av.r00f.us) |
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| 07:30.16 | v0lZy | i just moved some other lines away |
| 07:30.20 | v0lZy | let me review, one moment |
| 07:32.36 | v0lZy | appears to be the second sip packet |
| 07:33.29 | v0lZy | let me paste it for you hold on |
| 07:34.53 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 07:36.21 | v0lZy | seems to me that they agree on 180 |
| 07:37.06 | spicyramen_ | 1.Supported: replaces, timer |
| 07:37.06 | spicyramen_ | 2.Session-Expires: 180;refresher=uas |
| 07:37.09 | spicyramen_ | in 200 OK |
| 07:37.25 | spicyramen_ | Asterisk is the one in charge of sending the Session refresh |
| 07:38.43 | *** join/#asterisk hindi (~hindi@030-138-088-212.ip-addr.vsenet.de) |
| 07:40.49 | v0lZy | so it doesnt send it or what? |
| 07:41.03 | v0lZy | i mean, i have a session-minse=90 |
| 07:41.09 | v0lZy | so 180 should be within limits. |
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| 07:42.32 | spicyramen_ | what * version are you using ? |
| 07:44.20 | v0lZy | Asterisk 11.3.0 |
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| 09:14.49 | BeeBuu | hello,all |
| 09:15.54 | BeeBuu | i got this-->WARNING[12213][C-00000000] dsp.c: Can only calculate silence on signed-linear, alaw or ulaw frames :( |
| 09:16.10 | BeeBuu | what's problem in my * system? |
| 09:17.23 | BeeBuu | it was happened when another user join the conference. |
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| 09:24.30 | BeeBuu | anyone help me please? |
| 09:24.39 | BeeBuu | i got this-->WARNING[12213][C-00000000] dsp.c: Can only calculate silence on signed-linear, alaw or ulaw frames :( |
| 09:24.41 | BeeBuu | it was happened when another user join the conference. |
| 09:34.44 | Zogot | Any reason why the core show settings doesnt show the directories I specified in asterisk.conf? https://gist.github.com/zogot/b781f0ef713a9d860a81 |
| 09:35.14 | Zogot | It doesn correctly show me Running as User clearvox, Running as group clearvox when I go to the asterisk -r |
| 09:35.48 | Zogot | so i imagine its loading the file. i even specify it directly to this config file. but if i put my moh music in the default location only then does it find it |
| 09:42.05 | Zogot | hmm ok, by not defining the configuration file, it correctly loaded the directories |
| 09:42.48 | Zogot | when I ran config list it showed core /etc/asterisk/asterisk.conf which wasn't true, as when i showed core show settings, configuration file showed the specified location on startup |
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| 10:25.24 | file | Zogot, the (!) needs to be removed from directories |
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| 11:49.08 | yottanami | I just find out some one connected to my asterisk and used my phone and had some calls ! How can I check where was the problem ? |
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| 12:00.14 | yottanami | Is anyway to check logs which username was used ? |
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| 13:06.17 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 13:06.21 | FerForward | Hi, i have asteris 11.11 with freepbx 2.11 and custom contexts. i want to forward internal extension to mobile phones and show the caller id from the internal extensions. can anybody tell where i can find information about that? thanks |
| 13:06.57 | FerForward | i support two companies and i want that each forward shows the main number of each company. |
| 13:07.06 | *** join/#asterisk paolo_ (~smuxi@ip-62-143-36-139.hsi01.unitymediagroup.de) |
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| 13:10.40 | paolo_ | hi, i am experimenting with directrtpsetup in chan_sip. i recognize that asterisk is adding codecs to SDP. but this is wrong since asterisk won't translate. removing codecs should do no harm. How can I make asterisk not to add codecs. |
| 13:12.02 | paolo_ | other thing, what if asterisk didn't know the codec. It will remove it, right ? |
| 13:12.06 | [TK]D-Fender | paolo_: Your peer defines your codecs. And the order each side proposes will determine what gets negotiated and you could end up translating |
| 13:12.07 | FerForward | i'm not sure, but maybe: preferred_codec_only=yes |
| 13:12.42 | [TK]D-Fender | paolo_: And there is no "remove". It works on the codecs it's told to offer. Asterisk is NOT a "SIP server", or "proxy". It is a B2BUA |
| 13:13.02 | paolo_ | [TK]D-Fender: how could i and up transcoding ? i use directrtpsetup |
| 13:13.23 | [TK]D-Fender | paolo_: that's where POSSIBLE. But you are creting scenarios where it isn't |
| 13:13.28 | [TK]D-Fender | creating* |
| 13:13.49 | [TK]D-Fender | FerForward: When do you intend to undo this "forward"? |
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| 13:15.29 | paolo_ | [TK]D-Fender: this has nothing to do with beeing a b2bua or not. see yate. its also a b2bua but it can forward SDP to the other channel if both channel supoort it. |
| 13:15.35 | FerForward | depends on users. The users forward their numbers calling to *72 directly |
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| 13:15.58 | [TK]D-Fender | paolo_: That's Yate's infrastructure, not *'s. There is no "forward" |
| 13:16.36 | paolo_ | [TK]D-Fender: if directrtpsetup is used asterist must not add codecs. |
| 13:16.57 | [TK]D-Fender | FerForward: Your trunk needs to support setting the callerID, and your route & trunk can't be set in a way that overrides it. |
| 13:17.17 | *** join/#asterisk dfighter (~someone@arcemu/staff/dfighter) |
| 13:19.51 | [TK]D-Fender | paolo_: pastebin your peers & call with SIP debug |
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| 13:20.50 | FerForward | ok, thanks. i know that my trunk supports it, i'll need to unset the route and trunk cids |
| 13:36.32 | FerForward | i can't do it, and i need to goo to my kid. many thanks |
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| 13:39.28 | dan_j | Hi. In iax.conf, how does one specific a few different ip addresses in permit= ? |
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| 13:52.25 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 13:57.52 | paolo_ | [TK]D-Fender: http://pastebin.com/BsHhWqvJ |
| 13:58.30 | paolo_ | [TK]D-Fender: As you can see asterisk is not changing contact in SDP but it's adding codecs. |
| 13:58.49 | file | directrtpsetup is an experimental feature it may or may not work |
| 14:00.10 | paolo_ | file: i know. i am jsut saying adding codecs is always wrong when directrtpsetup is used. but asterisk adds codecs |
| 14:01.59 | paolo_ | file: doesn't chan_pjsip do the same ? should I use chan_pjsip ? |
| 14:02.15 | file | it does not have directrtpsetup |
| 14:03.45 | paolo_ | file: are you sure... "direct_media=yes ; Determines whether media may flow directly between; endpoints (default: "yes") |
| 14:03.48 | *** join/#asterisk igcewieling (~ewieling@172.sub-70-193-66.myvzw.com) |
| 14:03.58 | file | directrtpsetup and direct media are two different things |
| 14:04.13 | file | directrtpsetup occurs before a call is set up, direct media occurs after using re-invites |
| 14:04.29 | paolo_ | file: hmm... i thought pjsip doesn't use reinvites |
| 14:04.34 | file | it does. |
| 14:04.59 | igcewieling | I have an offtopic question: Has anyone seen on Polycom phones an issue where ** dialed at the beginning of a number is converted to a + ? |
| 14:05.03 | paolo_ | file: Ok. This is sad :-/ |
| 14:05.06 | mjordan | paolo_: I'm curious, why did you think it wouldn't use a re-INVITE? |
| 14:05.17 | mjordan | paolo_: And what is sad about a re-INVITE? |
| 14:05.53 | [TK]D-Fender | igcewieling: I know their dialplans support substitutions that may explain that... |
| 14:05.56 | paolo_ | mjordan: dunno. I was sure I read it somewhere that pjsip won't use re-invtes for direct_media |
| 14:06.09 | mjordan | how else would it inform a phone of where it needs to send media? |
| 14:06.15 | mjordan | And why do you think an INVITE request is a bad thing? |
| 14:06.31 | paolo_ | mjordan: i think you misread something |
| 14:06.41 | mjordan | "<paolo_> file: hmm... i thought pjsip doesn't use reinvites" |
| 14:06.46 | mjordan | "<paolo_> file: Ok. This is sad :-/" |
| 14:06.58 | paolo_ | mjordan: please stay in tzhe context |
| 14:07.08 | mjordan | I'll leave you to your musings then. |
| 14:07.20 | paolo_ | mjordan: we are talking about directrtpsetup |
| 14:07.29 | mjordan | which has nothing to do with a re-INVITE |
| 14:07.41 | paolo_ | mjordan: I know |
| 14:07.50 | paolo_ | mjordan: please read from the beginning |
| 14:07.58 | paolo_ | mjordan: i am sure you will understand |
| 14:08.00 | mjordan | I'll leave you to your musings then. Enjoy. |
| 14:08.27 | paolo_ | mjordan: i am true sorry if I somehow offend you. |
| 14:08.37 | paolo_ | truely |
| 14:09.03 | igcewieling | [TK]D-Fender: *nod* I'l trying to see what is different between this site and the other 50+ sites where this is not a problem. They should all be running the same configs. |
| 14:10.34 | [TK]D-Fender | igcewieling: converting 00 to + is a "standard" concept at least. I suspect something is there that does the same idea for **, but it wouldn't be hard-coded. Should be easy to find... either in provisioning, or on the phone itself |
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| 14:11.27 | igcewieling | [TK]D-Fender: exactly. |
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| 14:11.58 | igcewieling | [TK]D-Fender: but there is no character + in any of the config files |
| 14:12.11 | [TK]D-Fender | might be hex encoded, etc |
| 14:12.38 | [TK]D-Fender | just head right to the dialplan schema |
| 14:13.05 | igcewieling | [TK]D-Fender: unlikely. I might just sync the files from a working site. |
| 14:13.37 | [TK]D-Fender | igcewieling: I wouldn't toss out a proper investigation just to be sure.... |
| 14:13.44 | paolo_ | mjordan: what i want is, that asterisk is not in the media path. this is handled by re-invites in chan_sip. directrtpsetup "skips" this step. this is all I wan't. But it can't use it because asterisk adds codecs. that's why I ask if I could use chan_pjsip. thats all... sorry for the noise |
| 14:14.11 | *** part/#asterisk paolo_ (~smuxi@ip-62-143-36-139.hsi01.unitymediagroup.de) |
| 14:15.44 | [TK]D-Fender | Darn.. I just about had a solution for him... |
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| 14:19.05 | igcewieling | [TK]D-Fender: I would have remembered if I encoded something in the config. |
| 14:19.20 | igcewieling | I found an option which seems promising |
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| 15:15.02 | stevePearPear | Hi, can i check if thereâs anyway I can make a call without registration? Only authenticate on Call? |
| 15:15.49 | [TK]D-Fender | Registration has nothing to do with calling out. |
| 15:16.04 | [TK]D-Fender | ~sipregister |
| 15:16.04 | infobot | [~sipregister] SIP registration is to tell your provider what IP address & EXTEN to send INCOMING calls to. Some ITSPs let you use a fixed address or host rather than registering. Registration is NOT normally needed to PLACE calls, as those are typically auth'ed independently. Others accept unauth'ed calls once you are registered (saves on negotiation BW). |
| 15:17.03 | stevePearPear | hmm does this mean that I would send the authentication in my invite message? |
| 15:18.22 | [TK]D-Fender | registration does not affect your calling out a peer |
| 15:18.26 | [TK]D-Fender | they are not related |
| 15:18.37 | [TK]D-Fender | Whatever auth you have in ther ... you have in there. |
| 15:20.08 | stevePearPear | oh ya i just realized that! thanks! |
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| 15:27.46 | igcewieling | Sometimes I wonder if the ghost of the Marquis de Sade designed the SIP protocol. |
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| 15:30.31 | navaismo | Why? there is no self pleasure in it, or sodomia |
| 15:34.29 | igcewieling | Because no sane person would design SIP the way it is designed? |
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| 15:38.52 | mjordan | shrugs |
| 15:39.10 | mjordan | Any time you design a protocol, things start out nice and simple and sane. And then the real world barges in. |
| 15:40.14 | [TK]D-Fender | A gryphon built by committee... |
| 15:44.14 | mjordan | I prefer to focus on the parts that I like, and leave the ass end of the gryphon to others ;-) |
| 15:57.27 | Stefan27 | how can i get the 'Time: <int>' field to be included in peerstatus-events. judging from code in handle_response_peerpoke in chan_sip.c -- blob = ast_json_pack("{s: s, s: i}", "peer_status", s, "time", pingtime); ast_endpoint_blob_publish(peer->endpoint, ast_endpoint_state_type(), blob); -- shouldn't the time field be included? |
| 15:57.57 | *** part/#asterisk igcewieling (~ewieling@172.sub-70-193-66.myvzw.com) |
| 15:59.23 | Stefan27 | but i could modify the json_pack to include an additional {s:s} "cause" "foo" making the peerstatus event include a Cause: "foo" |
| 16:10.21 | Stefan27 | aha pingtime needs to be converted and formated to %s ... nvm |
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| 17:32.24 | opsview | any experts? |
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| 17:33.29 | opsview | hi |
| 17:33.43 | opsview | dead. |
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| 17:42.55 | navaismo | ~ask |
| 17:42.55 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 17:55.43 | [TK]D-Fender | and ..... crickets |
| 17:59.41 | *** join/#asterisk timahvo1 (~rogue@197.237.134.227) |
| 18:10.52 | salz212 | just wondering how can I manipulate state of remote-agent on some other server as trunk |
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| 20:05.20 | mjordan | salz212: I'm not sure what you mean by "manipulate the state". |
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| 21:14.53 | *** mode/#asterisk [+o file] by ChanServ |
| 21:18.23 | cablop | hello |
| 21:18.43 | cablop | is this a development channel or a support channel? |
| 21:19.18 | file | neither! it's a channel where people can talk about Asterisk - if that happens to be because there is an issue someone is trying to figure out, so be it |
| 21:22.18 | *** join/#asterisk bibz (~bibz@194-166-236-160.adsl.highway.telekom.at) |
| 21:22.18 | navaismo | so schrödinger-Asterisk? |
| 21:22.19 | bibz | hi guys |
| 21:22.45 | bibz | oh, I didn't even start explaining and @navaismo knew what I'm going to tell you |
| 21:23.32 | bibz | before compiling asterisk, I'll type in make menuselect.. the module func_odbc is checked (*), but after I compile asterisk (11.11), it won't list up on "module show"... |
| 21:23.53 | bibz | so its installed, but its not? some weird schrödinger-effekt? :P |
| 21:24.57 | navaismo | mein answer was for file |
| 21:25.21 | navaismo | but apply isnt it |
| 21:25.32 | bibz | I know, just wanted to crash into your conversation.. |
| 21:25.58 | mjordan | bibz: what does module show like func_odbc display? |
| 21:26.00 | cablop | nice |
| 21:26.40 | cablop | i'm trying to enable the AMI for queuemetrics |
| 21:26.50 | cablop | but i don't know how to |
| 21:27.13 | cablop | i edited the file according to the manual and still cannot access from queuemetrics |
| 21:27.58 | navaismo | can you login via Telnet |
| 21:27.58 | Nugget | telnet is eeeeeeevil! |
| 21:29.04 | navaismo | eeeeeeevil is goooood |
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| 21:31.27 | navaismo | " I am bad and that is good, I will never be good and that's not bad, there's no one I'd rather be than me" |
| 21:32.05 | bibz | any ideas? :) |
| 21:32.19 | navaismo | mjordan actually sent you an answer |
| 21:33.35 | navaismo | and consider that the answer from Mjordan its like when God answer to your prays... |
| 21:35.17 | mjordan | HA |
| 21:35.21 | mjordan | file would disagree. |
| 21:35.25 | bibz | oh, damnit. |
| 21:35.37 | bibz | I should buy some glasses |
| 21:35.41 | file | what am I disagreeing with now? |
| 21:35.45 | mjordan | Apparently I'm god |
| 21:36.09 | bibz | 0 modules loaded if I type in asterisk -rx "module show like func_odbc"... |
| 21:36.20 | mjordan | k, what does "module load func_odbc.so" display? |
| 21:36.38 | bibz | Unable to load module func_odbc.so |
| 21:36.38 | bibz | Command 'module load func_odbc.so' failed. |
| 21:36.52 | mjordan | you get no ERROR messages? |
| 21:36.55 | bibz | nope. |
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| 21:38.09 | bibz | but on make menuselect "odbc" is selected under the resource modules and dialplan functions categories.. |
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| 21:38.18 | navaismo | check your full log |
| 21:39.33 | navaismo | some modules cant be loaded without a valid config, enable the full log and take a look in the complete error messages |
| 21:39.44 | bibz | just did so, hold on just a sec |
| 21:40.19 | navaismo | mjordan: You are like GOD & file like st. Peter |
| 21:40.35 | file | I'm just a guy who likes music |
| 21:40.38 | bibz | http://pastebin.com/7xGABESi here you can see the content of /var/log/asterisk/full |
| 21:41.12 | navaismo | i cant see pastebins :'( domain blocked |
| 21:41.23 | bibz | [Aug 13 23:39:55] WARNING[28051] db.c: Couldn't execute statment: SQL logic error or missing database |
| 21:52.57 | navaismo | check your DB, maybe? |
| 21:52.58 | mjordan | hm. |
| 21:53.20 | *** part/#asterisk Yanxi (~procube@port-92-204-112-167.dynamic.qsc.de) |
| 21:53.25 | mjordan | func_odbc is probably bailing due to res_odbc puking. |
| 21:54.03 | mjordan | although that's actually db.c, which is going to be your astdb. |
| 21:54.10 | mjordan | So you've got a couple of things going wrong there. |
| 21:54.51 | mjordan | So yes. Step 1: fix your astdb. You may have a permissions issue between how your Asterisk process is being run and the permissions on the astdb |
| 21:58.58 | bibz | are there any good tutorials for how to install asterisk with odbc functionality?... |
| 21:59.18 | bibz | thanks for taking the time to answer my question |
| 22:05.40 | mjordan | bibz: I'd consult the book |
| 22:05.41 | mjordan | ~thebook |
| 22:05.41 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 22:05.55 | mjordan | ODBC integration is covered in there |
| 22:07.37 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 22:26.47 | *** join/#asterisk cablop (~kvirc@190.147.153.40) |
| 22:47.42 | bibz | aah its so hard to troubleshoot a 3000 lines dialplan.. |
| 22:48.16 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 22:52.06 | bibz | I have about 1000 calls per day and about 200 of them seem to can't get through and are listed as "NO ANSWER".. what could the problem be? |
| 22:52.14 | bibz | sometimes its even BUSY |
| 22:56.45 | bibz | probably network issues? |
| 23:07.56 | cablop | sigh |
| 23:08.00 | cablop | i've lost connection |
| 23:08.21 | cablop | but someone here knew the solution to the problem |
| 23:08.43 | cablop | they use FreePBX and that thing reloads the AMI when you add or modify users |
| 23:22.57 | *** part/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
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