| 00:00.05 | WIMPy | You shouldn't 'make samples'. |
| 00:00.30 | ruben23 | <PROTECTED> |
| 00:00.37 | ruben23 | conference i mean |
| 00:01.00 | WIMPy | Everyone calls the conference extension. |
| 00:01.27 | WIMPy | Or you transfer an existing call to that extension. |
| 00:02.13 | ruben23 | <PROTECTED> |
| 00:04.24 | WIMPy | I don't see a conference button. |
| 00:04.57 | ruben23 | WIMPy: on the home service- i see conference |
| 00:05.31 | WIMPy | Well, no idea then. |
| 00:06.10 | mzb | Qwell, np + done |
| 00:06.27 | mzb | (disabled infobot) |
| 00:06.44 | mzb | can't remember why it was enabled for #asterisk in the first place |
| 00:07.29 | *** join/#asterisk Witch_Doc (~WitchDoc@69.196.64.134) |
| 00:08.14 | Witch_Doc | has anyone successfully linked asterisk to a panasonic tde system? I'm stuck with this and any help would be appreciated |
| 00:08.36 | *** join/#asterisk mindCrime (~chatzilla@static-71-120-222-211.rlghnc.dsl-w.verizon.net) |
| 00:08.43 | WIMPy | Witch_Doc: Maybe you should tell us how you pan to link them. |
| 00:08.58 | WIMPy | plan |
| 00:09.22 | Witch_Doc | WIMPy the plan is to use asterisk as a conference bridge as well as being able to register sip extensions to the * and dial out via pbx |
| 00:09.26 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 00:10.11 | Witch_Doc | i have a sip gateway card setup in the pbx and can point it settings for registrar/proxy etc |
| 00:10.26 | WIMPy | How do you want to connect them? |
| 00:10.26 | WIMPy | Ah |
| 00:10.34 | Witch_Doc | not quite sure how to configure * though to |
| 00:11.23 | Witch_Doc | i' |
| 00:11.46 | Witch_Doc | i've sucessfully setup the panasonic pbx to accept registrations from sip phones and call other extensions |
| 00:12.25 | Witch_Doc | so i'm wondering can i have the pbx register with * and have * register a trunk with pbx as an extension |
| 00:12.29 | WIMPy | I'm pretty sure Asterisk will be able to talk SIP in whatever way might be necessary, but I'm sure others can tell you more about SIP than I can. |
| 00:13.06 | Serees | WIMPy no luck |
| 00:13.39 | *** join/#asterisk plundra (1000@v0.article.se) |
| 00:13.44 | WIMPy | Serees: Hmm. Funny. Anything changed from before? |
| 00:14.03 | Serees | creating debug now |
| 00:15.58 | WIMPy | Maybe the patch you used for dahdi wasn't the right one? But I'd expect that to cause earlier trouble. |
| 00:16.05 | *** join/#asterisk shapr (~shapr@nat/digium/x-oznihveurjkaorgu) |
| 00:16.51 | Serees | http://pastebin.com/Am33tuyy |
| 00:17.37 | Serees | from what i dnow there is no real patching done in dahdi... perhaps other than the required once |
| 00:18.31 | Serees | there are the patches done: |
| 00:18.33 | Serees | <PROTECTED> |
| 00:18.33 | Serees | <PROTECTED> |
| 00:18.33 | Serees | <PROTECTED> |
| 00:18.34 | Serees | <PROTECTED> |
| 00:18.34 | Serees | <PROTECTED> |
| 00:18.34 | Serees | <PROTECTED> |
| 00:18.34 | Serees | <PROTECTED> |
| 00:18.35 | Serees | <PROTECTED> |
| 00:18.46 | WIMPy | You need some knid of patch or 3rd party addon for a single port HFC card. |
| 00:19.27 | WIMPy | Please use pastebin next time. |
| 00:19.37 | WIMPy | What was that from? |
| 00:19.51 | Serees | its gentoo ebuild |
| 00:20.56 | WIMPy | It still can't get a TEI. |
| 00:21.22 | WIMPy | Lat's try something easy inbetween: Have you checked cabling? |
| 00:22.17 | Serees | well... no as it didn't change in between the working 1.2 and the non working 1.6... I havent been near the system since... |
| 00:22.27 | WIMPy | ok |
| 00:22.46 | Serees | at the moment checking cabling will be a bit difficult... but the thought has crossed my mind |
| 00:23.06 | Serees | but then I think it is strange that it worked on 1.2 and it wont work anymore in 1.6 |
| 00:23.20 | Serees | unless they have made the code less tollerant |
| 00:23.44 | WIMPy | It's digital. |
| 00:23.45 | Serees | but I doubt if that is even possible (don't have enough knowledge to judge about it) |
| 00:24.03 | WIMPy | So unless the gremlins pulled on the plug, that's probably not the issue. |
| 00:24.24 | Serees | hehe |
| 00:24.25 | Serees | lol |
| 00:24.54 | drmessano | I read that as "germans" first |
| 00:24.58 | drmessano | Achtung! |
| 00:25.30 | WIMPy | Ah, a faulty Stasi interface. |
| 00:25.49 | Serees | lol |
| 00:26.11 | Serees | I might be in germany atm... but i'm not german myself :p |
| 00:26.15 | WIMPy | Sorry, I meant Sina Box. |
| 00:26.46 | Serees | WIMPy sorry you lost me =s |
| 00:26.49 | WIMPy | Well, I am, but that happened, before I had a chance to do something baout it. |
| 00:26.58 | Serees | lol |
| 00:36.02 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 00:36.17 | Serees | WIMPy: some additional information |
| 00:36.30 | Serees | it seems that asterisk only see calls on chanel 1 |
| 00:36.45 | Serees | could it be that the chanels are misconfigured? |
| 00:37.02 | Serees | or doesn't that make any sense/difference |
| 00:37.03 | WIMPy | Huh? That doesn't make any sense to me. |
| 00:37.25 | WIMPy | If you were on the wrong D-Channel you wouldn't get anything sensible at all. |
| 00:41.29 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 00:42.27 | WIMPy | It looks to me as if you can't transmit. |
| 00:45.29 | *** join/#asterisk pc500 (~kvirc@AFS-Boise-Static-Customer-208-39-251-26.afsnetworks.com) |
| 00:45.44 | pc500 | can anyone recommend a USB headset for a soft phone? |
| 00:46.42 | WIMPy | pc500: Make sure it supports the 8000samples/s rate. |
| 00:47.37 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 00:48.52 | pc500 | WIMPy: - Anything that's good with g7222 (16000)? |
| 00:49.02 | pc500 | or is it just 8000 with an increased range |
| 00:49.03 | pc500 | hmmm |
| 00:49.23 | WIMPy | No, G.722 is 16000. |
| 00:49.45 | pc500 | I figured they'd use cheap PC mics which should do this |
| 00:50.27 | WIMPy | Yes, but the USB devices often can't change to a sample rate below 44100. |
| 00:51.03 | WIMPy | I don't know what the software situation is like now, but whe I tried such a combination, the result was pretty ugly. |
| 00:51.48 | pc500 | spec sheets seem useless in this matter :( |
| 00:52.17 | pc500 | one or two ears -- what do you recommend? |
| 00:52.49 | WIMPy | prefers two. But that's obviousely a personal choice. |
| 00:53.13 | Witch_Doc | two ears block more background noise |
| 00:53.25 | pc500 | I have an office with a door, if it matters. |
| 00:53.36 | pc500 | but mostly I'd use it when working at home. |
| 00:53.47 | Witch_Doc | gnnetcom has a usb adapter |
| 00:54.06 | Witch_Doc | then you can use any gn headset |
| 00:54.20 | pc500 | might be a good idea |
| 00:54.23 | pc500 | I was thinking http://www.newegg.com/Product/Product.aspx?Item=N82E16826265094&cm_re=usb_headset-_-26-265-094-_-Product or similar |
| 00:55.19 | *** join/#asterisk boch (~boch@190.220.65.19) |
| 00:55.47 | boch | hello all |
| 00:57.11 | boch | anybody knows how to remove php quick profiler from a2billing installation? i know is the proper place, where should i ask? |
| 00:57.11 | *** join/#asterisk JonnyD_work (~Jon@cpe-071-075-036-057.carolina.res.rr.com) |
| 00:57.43 | boch | is not i mean sorry |
| 00:58.28 | *** join/#asterisk svdasein (~dparker@dsl-63-249-115-213.dhcp.cruzio.com) |
| 00:59.52 | Serees | WIMPy and others... thanks for the help... i'm giving up for tonight... lets see if we have more luck tomorrow... Will try to get a tool installed so I can make phonecalls from here... maybe we get more inforrmation when I make outgoing calls from asterisk |
| 01:00.37 | WIMPy | 'channel originate' |
| 01:01.03 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 01:01.21 | Serees | later guy's |
| 01:01.24 | *** part/#asterisk Serees (~Serees@95.33.198.170) |
| 01:04.18 | svdasein | Hi - I'm extremely new to asterisk and am kind of trying to bootstrap my learning by using apstel visual dialplan to re-create an extensions.conf I've got that's known to work. I am doing ok, except I came up on this one thing that isn't obviously available in the ui. In this line: |
| 01:04.36 | svdasein | exten => <pattern>, n(bridged),Bridge(${DB_DELETE(gv_dialout/channel)}, p) |
| 01:04.59 | svdasein | I can't seem to find anything in visual dialplan that generates "Bridge()" commands |
| 01:05.29 | svdasein | is that command perhaps part of some other concept that they've have wrapped in the gui |
| 01:05.49 | svdasein | (sorry about asking here - I'm not aware of any irc for that product) |
| 01:06.18 | WIMPy | It's probably just not up to dated. |
| 01:06.39 | svdasein | so that's a relatively new command? |
| 01:06.48 | svdasein | (WIMPy) |
| 01:06.59 | WIMPy | yes |
| 01:07.03 | svdasein | ok - thanks! |
| 01:07.08 | WIMPy | They're called Applications BTW |
| 01:07.28 | svdasein | which - you mean "Bridge()"? |
| 01:08.06 | WIMPy | It's Applications not commands. |
| 01:08.14 | svdasein | ah ok - thanks again |
| 01:09.43 | svdasein | terminology in asterisk is definitely part of the challenge |
| 01:10.34 | WIMPy | Yes |
| 01:13.12 | *** part/#asterisk pc500 (~kvirc@AFS-Boise-Static-Customer-208-39-251-26.afsnetworks.com) |
| 01:40.24 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 01:53.14 | Freeaqingme | Is there an asterisk mailinglist one can spam? |
| 01:54.23 | Freeaqingme | s/spam/send legit email only to" |
| 02:16.18 | *** join/#asterisk GrizzlyAdams (~Grizzly@ip98-184-88-41.mc.at.cox.net) |
| 02:16.42 | GrizzlyAdams | :( i can't get google voice working with my asterisk install |
| 02:17.08 | GrizzlyAdams | it says its connected, but calls never ring through |
| 02:17.32 | elb | define "it" |
| 02:17.45 | elb | and calls never ring through in which direction |
| 02:17.59 | GrizzlyAdams | jabber show connections |
| 02:18.09 | GrizzlyAdams | incoming to asterisk from pstn |
| 02:18.34 | elb | if you 'jabber set debug on' at the asterisk prompt, do you see the incoming call? |
| 02:18.52 | GrizzlyAdams | nothing |
| 02:18.58 | elb | NB: I have Gtalk outgoing working 100%, but incoming fails about 2/3 times -- however, it always rings |
| 02:19.04 | elb | ok |
| 02:19.19 | elb | is your google voice account set to call your gtalk line? |
| 02:19.55 | GrizzlyAdams | yep |
| 02:20.07 | elb | pastebin your gtalk.conf and jabber.conf |
| 02:20.10 | elb | redact the passwords only |
| 02:21.38 | GrizzlyAdams | http://drunkencoders.com/pastebin/p/s8yaNP.html |
| 02:23.40 | elb | hmmm |
| 02:23.49 | elb | try adding status=xaway to the [asterisk] section in jabber.conf |
| 02:23.58 | elb | (I don't think that's it) |
| 02:24.15 | elb | and priority=1 |
| 02:26.38 | GrizzlyAdams | nuffin |
| 02:26.49 | elb | did you restart asterisk entirely? |
| 02:26.53 | GrizzlyAdams | yep |
| 02:26.55 | elb | ok |
| 02:27.15 | elb | (it's been my experience that restarting jabber and loading/unloading gtalk doesn't alwasy do it ... and is sometimes crashy-crashy) |
| 02:27.22 | elb | and you still get no indication that there's an incoming call? |
| 02:27.26 | elb | with jabber set debug on |
| 02:27.37 | GrizzlyAdams | no idications |
| 02:27.41 | elb | hum |
| 02:27.50 | GrizzlyAdams | i get presence tokens |
| 02:27.52 | elb | are you signed onto that gtalk account from another talk-capable client? |
| 02:28.27 | GrizzlyAdams | nope, but i can, and try calling it from my hardwired pstn line |
| 02:28.43 | elb | well, that's probably a good idea |
| 02:28.49 | elb | but I was actually thinking it might be breaking things, if you were |
| 02:29.01 | elb | I'm not sure how gtalk call routing works when there are multiple options |
| 02:29.11 | elb | but if it's like *normal* xmpp, it's ... complicated :-) |
| 02:35.59 | GrizzlyAdams | interestingly i get a ton of stuff if i sign in using pidgin while i have asterisk running |
| 02:36.11 | elb | yeah, you would |
| 02:36.20 | elb | xmpp is very chatty |
| 02:36.35 | elb | or do you mean you geta ton of stuff when you dial in? |
| 02:36.54 | GrizzlyAdams | nah, i get a ton of status about pidgin |
| 02:37.26 | GrizzlyAdams | and i am getting buddy notices on asterisk |
| 02:37.32 | elb | yeah |
| 02:37.37 | elb | becuase Pidgin adds buddies to the buddy list |
| 02:37.47 | elb | watch that, asterisk will delete your buddies under some circumstances |
| 02:37.54 | GrizzlyAdams | i mean when i sign in to my other gtalk account |
| 02:37.58 | elb | there's a jabber.conf option to tell it not to do that |
| 02:38.35 | elb | autoprune=no, I think |
| 02:40.35 | *** join/#asterisk JonnyD_work (~Jon@cpe-071-075-036-057.carolina.res.rr.com) |
| 02:41.09 | TeknoJuce | The best thing is just make a new gmail account just for asterisk then you dont have to worry about anything |
| 02:41.17 | GrizzlyAdams | right, i did that |
| 02:44.25 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 02:47.19 | GrizzlyAdams | lovely, now my cell number is locked to the pbx account :( |
| 02:47.29 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 02:48.51 | TeknoJuce | delete the account and readd a diff number |
| 02:54.57 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 02:55.01 | GrizzlyAdams | omg, if i call from my other account using the google voice plugin its ringing on the console |
| 02:55.14 | GrizzlyAdams | but asterisk is complaining about rtp missing |
| 02:57.01 | *** join/#asterisk manji (~manjiki@ppp-94-65-210-80.home.otenet.gr) |
| 02:57.05 | TeknoJuce | I had to add a patch to fix an rtp issues with google voice and my nortel i2004 phone (chan_unistim.c) |
| 02:57.28 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 02:57.38 | TeknoJuce | basically the phone would ring but if you picked up there was no audio |
| 02:57.54 | GrizzlyAdams | well its getting to my dialplan now |
| 03:03.06 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 03:05.13 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 03:09.26 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 03:11.13 | GrizzlyAdams | hrm, about 10 seconds before i get any sound |
| 03:13.27 | GrizzlyAdams | ok, looks like its something broken between google voice and google talk, cause i can use a google talk client to call in fine, but dialing from pstn doesn't work still |
| 03:15.33 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 03:15.46 | GrizzlyAdams | oooh, got it to notify now, but asterisk thinks its answered before it has actually :/ |
| 03:18.04 | *** join/#asterisk sourcode (~code@ppp-115-87-202-240.revip4.asianet.co.th) |
| 03:39.11 | TeknoJuce | did you figure it out |
| 03:39.53 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 03:40.14 | GrizzlyAdams | i think i'mma try the freeswitch interconnect method and see if i can get it going that way |
| 03:40.42 | GrizzlyAdams | i got it where it can get calls from the pstn, but the answer command never makes it to google |
| 03:41.09 | GrizzlyAdams | so its still ringing on the pstn side, but asterisk is running through the dialplan and eventually hangs up on the call |
| 03:43.11 | TeknoJuce | I just tried to call into my gv account havnt got it working yet will piss around with it |
| 03:43.28 | TeknoJuce | I get outbound just like you just fine |
| 03:44.32 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 03:44.48 | GrizzlyAdams | i haven't even tried outbound, since i have no other extensions hooked up yet. |
| 03:45.18 | GrizzlyAdams | sip is proving to be a pain to get working between my nexus-s and asterisk |
| 03:47.20 | TeknoJuce | are your 4 key files anything like this http://pastebin.com/YeEJDXHL |
| 03:47.32 | TeknoJuce | ignore the unistim.conf |
| 03:51.07 | TeknoJuce | hey elb what was the tutorial you said you used? |
| 03:51.20 | GrizzlyAdams | yep |
| 03:52.31 | GrizzlyAdams | question, you have call screening on or off? |
| 03:52.42 | GrizzlyAdams | cause the guide i read said to have it turned on or nothing would work |
| 03:53.28 | GrizzlyAdams | i'll try turning it on and seeing what happens |
| 03:53.32 | GrizzlyAdams | *off |
| 03:57.17 | TeknoJuce | are all your configs stripped down to the Bare Necessities? http://www.youtube.com/watch?v=9ogQ0uge06o |
| 03:58.19 | GrizzlyAdams | it should still be getting the Answer() command through to gtalk |
| 04:01.36 | TeknoJuce | also did you enable google talk in the phone options in the webpage voice settings |
| 04:01.48 | TeknoJuce | mine was turned off when I first added the phone number |
| 04:02.43 | GrizzlyAdams | yeah |
| 04:03.02 | GrizzlyAdams | i'm about to give up for the night |
| 04:03.32 | *** join/#asterisk jetlag (~jetlag@pool-173-61-216-136.cmdnnj.east.verizon.net) |
| 04:06.15 | TeknoJuce | options -> voice settings -> phone -> enable googlechat |
| 04:11.43 | tonsofpcs | options -> bug google to add feature -> SIP |
| 04:14.37 | GrizzlyAdams | options -> set fire to hair -> YES! |
| 04:15.59 | *** join/#asterisk jetlag (~jetlag@pool-173-61-216-136.cmdnnj.east.verizon.net) |
| 04:18.24 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 04:28.53 | sawgood | [2011-03-22 17:43:42] WARNING[14339] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 04:29.08 | sawgood | Is there anyway to figure out what caused this? |
| 04:29.22 | sawgood | This is a warning from the CLI logged to /var/log/asterisk/full |
| 04:29.49 | sawgood | It came up (back to back) for like 10 entries on the console |
| 05:01.28 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 05:03.19 | ChannelZ | That usually means the device is offline or plain invalid |
| 05:06.30 | *** part/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 05:20.43 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 05:20.49 | sawgood | ChannelZ: Thank you |
| 05:22.23 | sawgood | What is a good approach to take (for troubleshooting) when an incoming SIP trunk call to SIP end points keep ringing the phones (well after the incoming caller has hung up their phone)? |
| 05:23.05 | sawgood | The SIP phones do not stop ringing until someone picks one of them up (they might ring 10 more times after the caller to the trunk has hung up) |
| 05:23.07 | sawgood | strange! |
| 05:32.28 | shapr | sawgood: wireshark |
| 05:32.42 | shapr | see if the SIP phones are still being sent RINGING messages |
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| 06:09.27 | *** join/#asterisk Maxus2 (~Maxus@59.191.225.49) |
| 06:13.09 | Maxus2 | Hi People, im having trouble with regcontext for sip devices, if asterisk is reset, the sip devices fail to re-register themselves without a restart, any tips on how to get them to register without a restart? |
| 06:16.41 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
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| 06:35.25 | kaldemar | Maxus2: what do you mean by reset? do the registrations actually fail or do the phones even try to re-register? and what has regcontext to do with this? |
| 06:39.39 | Maxus2 | hi kaldemar, i have a sip phone connected to asterisk, in my sip.conf i have this entry: regcontext=sip_autoreg |
| 06:40.00 | Maxus2 | so when the phone connects and registers it automatically adds an entry to the dial plan: |
| 06:40.12 | Maxus2 | [ Context 'sip_autoreg' created by 'SIP' ] |
| 06:40.12 | Maxus2 | <PROTECTED> |
| 06:40.49 | Maxus2 | if the asterisk box is restarted, the dial plan appears blank and until the phoen is restarted it isn't updated. |
| 06:42.47 | Maxus2 | i would have thought that the phone when it updates it registration would have cause asterisk to re-add it to the dial plan |
| 06:42.52 | Maxus2 | or am i missing somthing here? |
| 06:43.33 | kaldemar | the phone probably doesn't even re-register. it doesn't know when asterisk is restarted. |
| 06:43.57 | *** join/#asterisk [netman] (~netman@144.Red-83-41-0.dynamicIP.rima-tde.net) |
| 06:44.48 | Maxus2 | but doesn't the phone check in on a regular basis? |
| 06:44.55 | kaldemar | Maxus2: expiry settings in sip.conf controll how often the clients should register. if that's not enough you'd need some way to tell the phones to register again. what phones are you using? |
| 06:45.26 | Maxus2 | linksys VoIP SPA941-AU |
| 06:45.44 | kaldemar | the phones don't actually check anything, they have an expiry timeout which is decided when they register and that is used as an interval for re-registrations. |
| 06:45.55 | kaldemar | then they just register again. |
| 06:46.22 | kaldemar | are you familiar with the sip notify command? |
| 06:47.19 | Maxus2 | nope |
| 06:47.24 | Maxus2 | just looking it up |
| 06:48.08 | Maxus2 | so your suggesting i send that out before restarting? |
| 06:48.12 | kaldemar | there are events for warm and cold reboots in sip_notify.conf by default for linksys phones. you might want to try those to reboot the phones remotely when asterisk is restarted. in CLI the command would be for example "sip notify linksys-warn-restart yourpeer". |
| 06:48.25 | Maxus2 | yeah that will be a no go for us |
| 06:48.43 | Maxus2 | i think we will just have to reduce the registration time out |
| 06:48.54 | kaldemar | why is that a no go? |
| 06:49.15 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 06:49.20 | Maxus2 | becuase the phones could be off site and run by our clients, thier not going to want to see thier phones magically restarting |
| 06:50.16 | kaldemar | is it worse than magically "not working"? i mean it should not be that often you restart a production pbx anyway. |
| 06:52.34 | Maxus2 | that is true, but ideally the phone should after say a minute kick back in |
| 06:53.38 | Maxus2 | so if i set the maxexpiry=3600 to say maxexpiry=60 that would cause them to re-register? |
| 06:54.49 | kaldemar | yes, it should. |
| 06:57.14 | Maxus2 | cool, thanks kaldemar. |
| 06:57.22 | Maxus2 | can i do the same for iax devices? |
| 06:58.47 | *** join/#asterisk jkroon (~jkroon@dsl-241-231-146.telkomadsl.co.za) |
| 06:59.18 | atan | Does anyone have details for how to setup RealTime with Asterisk 1.8.X? |
| 06:59.28 | atan | Does it expect the same table as 1.6.x? |
| 06:59.45 | Maxus2 | the script files are int he source |
| 07:00.04 | Maxus2 | thier in asterisk-1.8.2.4\contrib\realtime |
| 07:00.18 | Maxus2 | there is a folder for each db vendor |
| 07:00.58 | Maxus2 | i haven't used 1.6 to know if they are the same. |
| 07:05.53 | kaldemar | Maxus2: yes, you'll find similar settings in iax.conf. |
| 07:07.55 | Maxus2 | yep just found them, its wierd the phone still dont seemt o be registering themselves. |
| 07:08.50 | Maxus2 | im using realtim if that makes a difference |
| 07:11.55 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 07:12.26 | atan | I get " MySQL RealTime: Invalid database specified: asterisk (check res_mysql.conf)" |
| 07:12.31 | atan | I set dbname in res_mysql.conf |
| 07:12.39 | atan | realtime status mysql shows the same thing |
| 07:12.49 | atan | mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on (err 2002). Check debug for more info. |
| 07:13.00 | atan | Why would it not listen to my dbuser, dbname parms? =\ |
| 07:13.07 | atan | adds quotes to the name and such |
| 07:13.52 | atan | Nah, same crap. MySQL RealTime: Invalid database specified: asterisk |
| 07:14.18 | kaldemar | atan: what do you have as the DB name in extconfig.conf? |
| 07:15.08 | SiNGLer | atan: in extconfig specify "general" instead of "asterisk" |
| 07:15.41 | kaldemar | Maxus2: you'll see the current value with "sip show settings" in CLI. it may of course be that the phones don't care about what asterisk says. |
| 07:18.49 | atan | SiNGLer, I've changed this and issued 'reload' |
| 07:19.22 | atan | Still showing the 'asterisk' name. Odd. |
| 07:19.30 | atan | Perhaps I can change it to my db name in both places and see what happens |
| 07:20.01 | SiNGLer | atan: show your extconfig and db config |
| 07:20.13 | SiNGLer | (pastebin) |
| 07:22.17 | atan | http://pastebin.com/i2S6WQYC |
| 07:22.20 | atan | SiNGLer, thanks |
| 07:22.46 | atan | You'll notice the number prefix there, _secure. I just forgot to swap that out is all. |
| 07:22.58 | atan | Those are all the same in each file. |
| 07:23.20 | atan | http://pastebin.com/tnEFqTWL even |
| 07:24.27 | SiNGLer | well it seems to be ok, try restarting asterisk |
| 07:24.48 | atan | SiNGLer, you wouldn't be aware of a command to tell me if there are any active calls? |
| 07:24.58 | atan | Or perhaps a graceful restart which waits for all current calls to end? |
| 07:25.38 | SiNGLer | "core show channels" and "core restart when convenient", autocomplete last word, I may be misspelled it :) |
| 07:25.49 | atan | Sure. |
| 07:26.01 | atan | Okay I restarted anyway - current calls might have dropped but it's like 4am.. I doubt there were any. |
| 07:26.07 | atan | Still have the issue, it's showing MySQL RealTime: Failed to connect database server asterisk on (err 2002). |
| 07:26.23 | atan | Interesting. |
| 07:26.26 | atan | general configured for asterisk on socket file with username asterisk. |
| 07:27.19 | SiNGLer | I do not see general in your pasted configs. try using your profile on general :) |
| 07:28.11 | *** join/#asterisk juliocesarlhg (~jcesarg@190.234.250.59) |
| 07:29.19 | atan | I changed all the secure out to general but now I get other errors.. hmm, interesting |
| 07:29.32 | atan | <PROTECTED> |
| 07:29.40 | juliocesarlhg | who knows about mgcp.conf? |
| 07:29.47 | juliocesarlhg | i want to register a gateway |
| 07:31.05 | SiNGLer | atan: from where did "user" come? |
| 07:31.15 | juliocesarlhg | mgcp gateway |
| 07:32.43 | atan | SiNGLer, my only guess might be extconfig.conf which has dbuser and dbname |
| 07:33.53 | SiNGLer | can you show current config? |
| 07:35.33 | atan | SiNGLer, which file? I'd be happy to |
| 07:35.52 | atan | I wonder why it's not picking up anything from res_mysql.conf |
| 07:35.55 | atan | Nothing, nada |
| 07:36.52 | SiNGLer | maybe you did migrate mysql config from 1.4? |
| 07:37.35 | atan | Nah it's new |
| 07:37.44 | atan | But res_mysql.conf didn't exist, I had to create it |
| 07:37.50 | atan | Ah! Perhaps the wrong owner of the file. |
| 07:38.51 | atan | Darn. Nope. |
| 07:40.01 | SiNGLer | show your extconfig and res_mysql |
| 07:40.16 | SiNGLer | and what version of asterisk do you use? |
| 07:40.41 | atan | 1.8.1 |
| 07:41.40 | atan | SiNGLer, http://pastebin.com/7qnKKTud |
| 07:41.48 | atan | I've stripped the pass and host is all |
| 07:43.48 | atan | I'm wondering more about this "general configured for asterisk on socket file with username asterisk." line it's feeding me |
| 07:43.50 | SiNGLer | hm, maybe asterisk does not like underscore or digits in username? I never tried this form of username/sbname :) |
| 07:43.53 | atan | Where on earth is it getting that username. |
| 07:44.20 | SiNGLer | (09:29:32) atan: load_mysql_config: MySQL RealTime: No database user found, using 'asterisk' as default. |
| 07:44.41 | SiNGLer | maybe from here it gets "asterisk" username |
| 07:46.01 | SiNGLer | currently I do not have 1.8 setup, so I cannot test it |
| 07:47.37 | atan | Is there a realtime config somewhere I didn't see? |
| 07:48.45 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:211d:1ceb:94f2:b76f) |
| 07:49.10 | SiNGLer | I will try to install 1.8 into virtual machine, will try to check :) |
| 07:50.27 | atan | where are the default conf files found? |
| 07:50.31 | atan | Perhaps there are examples in there I can go off |
| 07:51.00 | SiNGLer | everything is in /etc/asterisk |
| 07:51.15 | SiNGLer | or you can look at source |
| 07:51.28 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 07:52.03 | SiNGLer | <PROTECTED> |
| 07:53.49 | atan | Upgrade to asterisk 1.8 res_mysql.conf has been changed to res_config_mysql.conf. |
| 07:53.49 | atan | sec |
| 07:54.29 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:55.23 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 07:59.23 | *** join/#asterisk [netman] (~netman@128.Red-80-39-55.staticIP.rima-tde.net) |
| 08:00.07 | atan | w00t |
| 08:00.21 | atan | SiNGLer, the filename changed but it didn't in the error thing. |
| 08:01.50 | SiNGLer | so now it works? :) |
| 08:02.01 | atan | Yep! |
| 08:02.04 | atan | Filename was wrong. |
| 08:02.10 | atan | res_mysql.conf is now res_config_mysql.conf. |
| 08:02.20 | atan | Now I just need to deal with loading in the sip people :D |
| 08:02.28 | SiNGLer | :) |
| 08:02.30 | atan | Need to figure out the database table scheme |
| 08:02.38 | *** join/#asterisk Tim_Toady (~moi@79.103.49.227.dsl.dyn.forthnet.gr) |
| 08:03.25 | SiNGLer | I thought that name changed on 1.6.2, so I checked there, and found res_mysql, so probably I was mistaken for trunk when I last time tried it |
| 08:03.31 | SiNGLer | DB scheme should be in source |
| 08:04.27 | SiNGLer | contrib/realtime/mysql |
| 08:07.25 | atan | "MySQL RealTime: Failed to query database. Check debug for more info." |
| 08:07.33 | atan | Interesting. Where's debug hiding at? |
| 08:07.41 | SiNGLer | <PROTECTED> |
| 08:08.13 | *** join/#asterisk appel11 (~root@78-22-118-226.access.telenet.be) |
| 08:08.14 | SiNGLer | I'd use "full => notice,warning,error,verbose,debug" |
| 08:08.47 | SiNGLer | and would check /var/log/asterisk/full |
| 08:09.09 | juliocesarlhg | who uses mgcp.conf??? |
| 08:09.11 | juliocesarlhg | please |
| 08:10.43 | appel11 | Hello, Is it possible to change the "To:" in the sip header, so it's differend of that of the host= settings? |
| 08:11.47 | atan | err, MySQL RealTime: Failed to query database. Check debug for more info.. full isn't showing why though |
| 08:12.12 | SiNGLer | atan: you uncommented it and reloaded logger? |
| 08:12.28 | SiNGLer | appel11: try playing with outboundproxy setting |
| 08:13.16 | appel11 | I've tried, call is canceled because of a loop detection |
| 08:14.48 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:15.03 | atan | My debug log is empty. Err. |
| 08:15.10 | atan | This stinks! lol. and "sip reload" |
| 08:15.23 | atan | Sorry, "sip reload" is gone |
| 08:15.33 | SiNGLer | you are checking file named "full"? |
| 08:16.02 | atan | full just shows res_config_mysql.c: MySQL RealTime: Failed to query database. Check debug for more info. |
| 08:16.11 | atan | There's no details on what col it was expecting but didn't find or anything |
| 08:17.21 | atan | Add set debug = 999 and now we have details |
| 08:17.25 | atan | One sec while I skim over them :D |
| 08:18.32 | atan | MySQL RealTime: Query Failed because: Unknown column 'category' in 'field list' |
| 08:20.11 | *** join/#asterisk lftsy (~lftsy@pul-lav-fw-so-01-x1.vtxnet.net) |
| 08:21.19 | *** join/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
| 08:21.52 | Get_The_Fish | yo, is the yum system for packaging asterisk @asterisk.org still be maintained, does anyone know? |
| 08:22.04 | Get_The_Fish | cause it doesn't seem to be working |
| 08:24.11 | atan | No such command 'sip show peers' |
| 08:24.25 | Get_The_Fish | chan_sip is not installed atan |
| 08:24.33 | atan | Get_The_Fish, but it is. |
| 08:24.38 | atan | It's in use now. |
| 08:25.14 | Get_The_Fish | do any of the sip commands work? |
| 08:26.13 | Get_The_Fish | nevermind, there is nothing there for RHEL.... hmmm |
| 08:27.07 | atan | Uht-oh! |
| 08:27.17 | atan | All sip stuff is gone now. What the hell did I touch. |
| 08:27.31 | Get_The_Fish | module probably failed to load |
| 08:28.38 | atan | Stopped / started it and we are good now. |
| 08:28.57 | Get_The_Fish | check your error logs than, cause it deadlocked |
| 08:29.06 | Get_The_Fish | (probably) |
| 08:29.12 | Get_The_Fish | what version are you on? |
| 08:29.16 | atan | I was messing with the realtime loading no doubt |
| 08:29.18 | atan | 1.8.1 |
| 08:29.30 | atan | I am trying to get my SIP peers to load off a MySQL table |
| 08:29.36 | atan | well, my SIP everything really. |
| 08:29.40 | Get_The_Fish | yeah ok that could do it if you had some garbage data in there or something |
| 08:30.53 | appel11 | hmm, SiNGLer any idea how to fix a loop detection? or let asterisk ignore this? |
| 08:31.40 | SiNGLer | appel11: sorry, no ideas from me.. |
| 08:33.28 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:36.27 | atan | Well bahumbug. |
| 08:36.32 | kaldemar | Get_The_Fish: it is. |
| 08:36.42 | atan | I ahve the SQL bit connecting but it's not grabbing SIP peers from the database |
| 08:37.11 | SiNGLer | any errors? |
| 08:38.17 | Get_The_Fish | man, I am going to have to make my own RHEL 6 yum repo |
| 08:38.22 | Get_The_Fish | that suuuuuuuuucks |
| 08:38.29 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:38.37 | *** join/#asterisk zkn (~zkn@195.222.14.202) |
| 08:39.24 | Get_The_Fish | cause god only knows when centos 6 is going to be out |
| 08:41.12 | zkn | Hi, could anyone give some insight into the following error messages: |
| 08:41.12 | zkn | ERROR[11775]: netsock2.c:245 ast_sockaddr_resolve: getaddrinfo("82.207.44.2", "27277rinstance=e40ced6755274850", ...): Servname not supported for ai_socktype |
| 08:41.12 | zkn | WARNING[11775]: chan_sip.c:9318 set_destination: Can't find address for host '82.207.44.2:27277rinstance=e40ced6755274850' |
| 08:41.37 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 08:45.31 | juliocesarlhg | help with mgcp configuration |
| 08:46.09 | Get_The_Fish | zkn, google the phrase "Servname not supported for ai_socktype" |
| 08:46.25 | zkn | doing it right now |
| 08:46.30 | Get_The_Fish | that is a common error message, as it originates the from the OS's network stack. |
| 08:46.37 | Get_The_Fish | everything is saying /etc/services |
| 08:46.51 | Get_The_Fish | full volume? |
| 08:49.36 | zkn | i don't even know if I had or am supposed to have /etc/services with OpenSUSE |
| 08:49.47 | *** part/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
| 08:50.04 | appel11 | SiNGLer: playing with outboundproxy did the trick, sadly It didn't fix my billing issue with the provider. Thanks for the tip! |
| 08:51.02 | SiNGLer | np |
| 08:55.37 | *** join/#asterisk jacc0 (~jacco@D522448D.static.ziggozakelijk.nl) |
| 08:55.41 | jacc0 | hi all |
| 08:55.46 | jacc0 | good morning :) |
| 08:56.31 | jacc0 | I have a question about async AGI using a PHP script: is it posible and how? |
| 08:57.32 | jacc0 | this is what I have in my dialplan now : AGI(/etc/asterisk/scripts/EraseMessage.agi,${CallID}-${shiftID}-${followupID}-${actionID}) |
| 08:57.41 | jacc0 | can I makeit run async? |
| 09:01.08 | jacc0 | I used to use: exec(php eraseMessage.php ${CallID}-${shiftID}-${followupID}-${actionID} &) |
| 09:01.15 | jacc0 | is that the best way? |
| 09:03.57 | kaldemar | jacc0: app Exec executes a dialplan application, not a shell command. use app System or func SHELL to run shell commands. those have nothing to do with AGI then. |
| 09:04.28 | jacc0 | okay, but no way to run agi async? |
| 09:04.47 | jacc0 | (i ment system in my example) |
| 09:04.57 | jacc0 | (my mistake) |
| 09:05.53 | *** join/#asterisk MrTelephone (~MrTelepho@h697179-171.picriverisp.net) |
| 09:06.48 | MrTelephone | anyone elses alarm clock going off? |
| 09:07.01 | zkn | not mine |
| 09:08.25 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 09:09.57 | MrTelephone | how far is digium from orlando |
| 09:11.03 | jacc0 | I read: "Async AGI Introduced in Asterisk 1.6, allows asynchronous AGI scripting" on the internet everywhere |
| 09:11.07 | jacc0 | but no example |
| 09:11.18 | jacc0 | can anyone point me to an example? |
| 09:11.40 | MrTelephone | I don't even know what AGI is |
| 09:14.56 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 09:14.58 | schmidts | good morning |
| 09:15.20 | jacc0 | good morning |
| 09:15.31 | jacc0 | do you know anything about async agi? |
| 09:16.12 | kaldemar | jacc0: "core show help agi exec" and "manger show command AGI" |
| 09:17.02 | jacc0 | ty |
| 09:17.02 | MrTelephone | anonymous www-authenticate messages. How does asterisk know how to authenticate? uri suboption in the WWW-authenticate header? |
| 09:18.34 | MrTelephone | AGI looks pretty cool. What do people use it for? |
| 09:19.45 | schmidts | MrTelephone for nearly everything. IMHO you can solve much of it using a good dialplan and maybe some agi but that depends on your needs |
| 09:20.55 | MrTelephone | I see that there is speech recognition in 1.6. That would be nice |
| 09:21.11 | MrTelephone | "Please say or key in the callers name" |
| 09:22.24 | *** join/#asterisk jploh (cbb18d7a@gateway/web/freenode/ip.203.177.141.122) |
| 09:24.38 | jploh | Can anyone help me with my * on a vm, 256MB 64-bit v1.6.1.1? I get a segfault at rtp.c:1458 |
| 09:25.02 | jploh | We're using skypeforasterisk as well |
| 09:27.29 | MrTelephone | yeah I wanted to try that but I don't think I can get enough users using it |
| 09:30.58 | MrTelephone | does it work good jploh? |
| 09:31.52 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 09:31.57 | MrTelephone | jploh, what is in rtp.c at line 1458? |
| 09:32.37 | jploh | MrTelephone: if (rtp->rtcp->rxjitter_count == 1) |
| 09:34.57 | MrTelephone | is that an error you get when you compile? |
| 09:35.54 | jploh | When I start a call |
| 09:36.19 | jploh | BTW, modules are not autoload |
| 09:36.49 | jacc0 | I realy don't understand the help I get from "core show help agi exec" |
| 09:37.01 | jacc0 | Usage: agi exec <channel name> <app and arguments> [id] |
| 09:37.15 | MrTelephone | jploh, did you try a 32-bit OS? |
| 09:37.29 | jacc0 | can anyone give an example on how to implement it into a dialplan? |
| 09:37.59 | jploh | MrTelephone: yes |
| 09:38.20 | jploh | MrTelephone: and error was the same |
| 09:39.08 | MrTelephone | jacc0, did you check http://www.voip-info.org/wiki/view/Asterisk+AGI already? |
| 09:40.11 | MrTelephone | jploh, do you think it's vm related? If that version of asterisk with your configuration works good on a standalone box then I would have to say it is a clock/timing issue |
| 09:41.30 | MrTelephone | i see posts of clock drift on virtual machines so I don't know how stable it would be using asterisk on one. I never tried. I used some 64-bit kernels on Microsoft Hyper-V with very little success. That's all I know. |
| 09:41.37 | jploh | MrTelephone: okay, it seems to work with 1.6.2.0 |
| 09:41.56 | MrTelephone | but you have to use 1.6.1.1? |
| 09:42.02 | jploh | MrTelephone: it worked on another extension though |
| 09:42.18 | MrTelephone | that is weird |
| 09:42.25 | jploh | MrTelephone: no version restriction. just have to use it with this vm for now |
| 09:42.51 | MrTelephone | what is your host operating system? I'm interested how your running asterisk as a virtual machine? |
| 09:43.25 | jploh | MrTelephone: not sure. it's on slicehost |
| 09:43.42 | MrTelephone | so you dial one extension and it seg faults and the others are ok? |
| 09:43.55 | jacc0 | MrTelephone: there is nothing there about agi exec |
| 09:45.04 | MrTelephone | pisspoor documentation on agi exec eh |
| 09:46.36 | MrTelephone | Something like this Example: $AGI->exec('Dial', 'Zap/g2/8005551212'); isn't enough to go on? |
| 09:47.02 | jacc0 | nope, that is not what i'm looking for |
| 09:47.41 | jacc0 | i want to run an AGI script async from the dialplan; I don't want to run a dialplan application from a AGI script async |
| 09:49.01 | jploh | MrTelephone: first it was to skype to a sip channel to a phone which worked. then skype to another * box via sip |
| 09:49.50 | jploh | Did another test call and it passed. I hope it was just with the version. |
| 09:49.57 | jacc0 | anyhere who can help me with an example? |
| 09:51.09 | jploh | jacc0: example for what? |
| 09:51.31 | MrTelephone | jacc0, explain to me what async agi is as opposed to regular agi |
| 09:51.49 | zkn | regarding with my above excerpts: |
| 09:51.49 | zkn | ERROR[11775]: netsock2.c:245 ast_sockaddr_resolve: getaddrinfo("82.207.44.2", "27277rinstance=e40ced6755274850", ...): Servname not supported for ai_socktype |
| 09:51.49 | zkn | WARNING[11775]: chan_sip.c:9318 set_destination: Can't find address for host '82.207.44.2:27277rinstance=e40ced6755274850' |
| 09:51.49 | zkn | does this even look fine? i mean, especially that "27277rinstance=e40ced6755274850" part? could there be a parsing error? |
| 09:52.51 | jacc0 | mrTelephone: an asyn agi will not wait for the agi script to finish in comperason to a normal AGI script |
| 09:53.13 | MrTelephone | I see |
| 09:53.58 | MrTelephone | i mean im searching async agi but I assume you did as well |
| 09:53.59 | jploh | jacc0: Can't you fork out from a fastAGI? |
| 09:54.45 | zkn | oddly enought, this happens only with one specific outgoing trunk, others on the same server are fine when outbound calls are made |
| 09:55.16 | MrTelephone | zkn: check your config file there is probably a mistake in it |
| 09:58.19 | zkn | what i have changed was nat=yes from nat=no, which brought the outbound trunk out of the UNREACHABLE state, so it was possible to call again but logs show these errors in the beginning and the end of every call |
| 09:59.22 | zkn | i'll try to narrow down the config for this trunk context, see if that helps |
| 09:59.28 | jacc0 | damn, I'll just use app_originate to start an async call and run tha AGI from there |
| 09:59.42 | MrTelephone | i just meant check the config to see if you have an extra comma or something weird in the peer part |
| 10:00.22 | zkn | MrTelehpone, nope, no commas or other accidental characters |
| 10:00.42 | MrTelephone | the calls work but you get a warning? |
| 10:00.47 | zkn | yes |
| 10:00.53 | zkn | which don't look too assuring |
| 10:01.49 | MrTelephone | might be a small programming mistake? are you using 1.8.3 or something? |
| 10:01.58 | zkn | yes |
| 10:02.06 | MrTelephone | i would have never guessed |
| 10:02.22 | zkn | 1.8.3 it is indeed |
| 10:02.26 | MrTelephone | check chan_sip.c and find out why set_destination is even called |
| 10:02.35 | MrTelephone | haha |
| 10:02.40 | MrTelephone | start making a patch |
| 10:02.40 | zkn | ummm.. |
| 10:02.51 | zkn | that's out of my league |
| 10:03.53 | MrTelephone | It looks like they have a variable out of place when they call the getaddrinfo() sub |
| 10:04.02 | jacc0 | the problem with app_originate is that it has an hardcoded timeout of 30secondes that makes it kind of useless; Iguess I'll have to hack the asterisk source and make it a 36000seconds timeout |
| 10:04.04 | MrTelephone | or they are missing a comma |
| 10:04.59 | zkn | hmm...i wonder how other trunks are fine then |
| 10:05.03 | MrTelephone | netsock2.c:245 ast_sockaddr_resolve: getaddrinfo("82.207.44.2", "27277rinstance=e40ced6755274850", ..) probably should read netsock2.c:245 ast_sockaddr_resolve: getaddrinfo("82.207.44.2", "27277","rinstance=e40ced6755274850", ...) |
| 10:05.13 | MrTelephone | but it's just a stab in the dark |
| 10:05.31 | zkn | yes, that's what i think, too...parsing error of some sort |
| 10:05.45 | MrTelephone | thats why i thought it was a config issue cause of it only working on one trunk |
| 10:06.17 | MrTelephone | unless that trunk is sending a sip packet that is invoking something else that isn't crucial to the call but is generating an error |
| 10:06.44 | MrTelephone | sip debug the trunk peer and see if there is anything different from the other trunks |
| 10:08.04 | MrTelephone | zkn, is that the ip and port of the trunk? |
| 10:08.32 | zkn | nope |
| 10:08.43 | MrTelephone | why is it trying to resolve that then |
| 10:08.51 | zkn | beats me :) |
| 10:09.05 | MrTelephone | a sip redirect or something |
| 10:09.40 | zkn | i was looking through /etc/services and that post was not listed on it.. thought this could also affect it... but the fact that this IP is unknow, is making things a little bit stranger |
| 10:09.54 | zkn | post=port |
| 10:10.28 | zkn | i could contact the ITSP and ask what IP is it |
| 10:10.41 | MrTelephone | sip debug one call to the trunk and it should be in the sip messages somewhere |
| 10:10.52 | zkn | ok |
| 10:12.44 | MrTelephone | see anything? |
| 10:12.56 | zkn | is there a way to silnece everything else byt the peer debug info ? :) |
| 10:13.06 | zkn | damn typos.. |
| 10:14.33 | MrTelephone | sip debug peer <trunkpeername> |
| 10:14.34 | MrTelephone | ? |
| 10:14.52 | MrTelephone | is there a lot calls going out on that trunk? |
| 10:14.59 | MrTelephone | oops |
| 10:15.00 | zkn | not many... |
| 10:15.01 | MrTelephone | i dunno |
| 10:15.08 | MrTelephone | i misread your question |
| 10:15.10 | zkn | the info is flowing tho |
| 10:15.25 | MrTelephone | see anything there with 82.207 in it? |
| 10:15.26 | MrTelephone | lol |
| 10:15.33 | zkn | will search |
| 10:17.32 | *** join/#asterisk cjk (~cjk@85.93.217.128) |
| 10:17.41 | cjk | hi, how can i set nat=yes for autocreated peers? |
| 10:19.32 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 10:20.02 | *** join/#asterisk sgimeno (~chatzilla@163.117.206.10) |
| 10:44.39 | jacc0 | cjk: try adding nat=yes to [default] in sip.conf |
| 10:45.46 | jacc0 | cjk: i'm not sure what you mean with "autocreated peers" but i guess you allow peers without authentication |
| 10:46.42 | jacc0 | peers that do not have an account are have an account that is missing the nat=yes will use the settings from [default] in your sip.conf |
| 10:46.53 | jacc0 | are=or |
| 10:47.02 | cjk | jacc0, ok thanks |
| 10:48.41 | jacc0 | yw |
| 10:50.02 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85de.bcn.adamo.es) |
| 10:54.08 | cjk | is there a way to define a fix jitter buffer per peer? |
| 10:55.19 | kaldemar | there's no [default] in sip.conf. |
| 10:56.40 | zkn | [general] |
| 11:00.13 | *** join/#asterisk Faithful (~Faithful@202.189.73.144) |
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| 11:17.32 | jacc0 | sorry, [general] ios what I ment |
| 11:17.35 | jacc0 | :) |
| 11:17.40 | *** part/#asterisk kaii (~kh@ciphron.de) |
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| 12:00.42 | kleszcz | ERROR[4486]: chan_sip.c:16860 handle_request: Dropping this SIP message with Call-ID '52532868553549819BBF2281AFCD10D8', it's incomplete. |
| 12:02.44 | cobra2 | I was just told by a friend that some devs were working on source for getting text message support in asterisk outside of the context of a call. Is this info correct? |
| 12:08.55 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
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| 12:19.17 | saxa | hi,= a simple question, the context= option in iax.conf means when box a registers to box b will have calls processed in this context ? |
| 12:19.25 | blitzrage | no |
| 12:19.36 | blitzrage | registration does not control call flow or where it enters in the dialplan |
| 12:19.41 | blitzrage | the authentication block does that |
| 12:20.22 | blitzrage | registration *only* tells the other end where your asterisk system resides on the network |
| 12:20.50 | saxa | ok |
| 12:21.14 | saxa | so i need to have one register statement for each box |
| 12:22.03 | saxa | and on each box I need to have also the [boxa] on boxb and [boxb] on box a ? |
| 12:22.10 | saxa | in iax.conf |
| 12:22.35 | saxa | where I put the username and password for authenticate them |
| 12:22.54 | *** part/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 12:23.20 | saxa | register => boxb:pass@boxa.com goes into iax.conf on box b , correct ? |
| 12:23.21 | blitzrage | saxa: you need to read some documentation |
| 12:23.26 | blitzrage | ~newbook |
| 12:23.26 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 12:23.39 | saxa | i have done that, and it confuses me a little bit |
| 12:24.35 | saxa | is it enough that i register the boxb to boxa to make calls in both ways correct ? |
| 12:24.39 | blitzrage | no |
| 12:24.51 | saxa | so both need to register one to each otehr ? |
| 12:24.51 | blitzrage | I just said that doesn't authenticate calls, it only tells the other box where you are |
| 12:24.57 | blitzrage | yes, that is where you'll start |
| 12:25.03 | blitzrage | you still need: |
| 12:25.05 | saxa | ok |
| 12:25.06 | blitzrage | [boxa] |
| 12:25.08 | blitzrage | type=peer |
| 12:25.11 | blitzrage | context=incoming |
| 12:25.13 | blitzrage | etc... |
| 12:25.14 | saxa | yes i have that |
| 12:25.24 | blitzrage | sorry, I have to run off and work on documentation |
| 12:25.34 | saxa | thats on box b in any case correct ? |
| 12:25.39 | kaldemar | if a doesn't register to b, b does not know where it is. no matter if b is registered to a or not. |
| 12:25.46 | saxa | blitzrage: ok, dont worry, thx anyway |
| 12:25.56 | blitzrage | don't worry, I'm not worried :) |
| 12:26.05 | saxa | great :) |
| 12:26.49 | saxa | kaldemar: ok, ths means I should have on both boxes in iax.conf the register statements from a to b and from b to a ? |
| 12:27.01 | kaldemar | saxa: exactly. |
| 12:28.02 | saxa | and on the oposite side of each box I should have in iax.conf [boxa] type=peer and [boxb] type=peer |
| 12:28.15 | saxa | with all other options |
| 12:28.26 | saxa | is that correct ? |
| 12:28.49 | saxa | so let me resume on pastebin |
| 12:28.51 | kaldemar | yep. |
| 12:29.45 | kaldemar | when you have register => user:pass@address, the other end at "address" needs a corresponding [user] with secret=pass. |
| 12:31.28 | saxa | http://pastebin.com/V3gUJ6qJ |
| 12:31.34 | saxa | is that correct ? |
| 12:32.01 | zkn | you can also use peer contxt with host= interconnect box a and box b |
| 12:32.19 | kaldemar | saxa: yes. |
| 12:32.23 | saxa | kaldemar: ok for the user thing I know they need to match |
| 12:32.54 | saxa | ok, so now, can both have the same context=incoming ? |
| 12:33.12 | saxa | zkn: i dont get you ? |
| 12:33.24 | kaldemar | saxa: they can be whatever you want. as long as you have a context by that name in your dialplan. |
| 12:34.20 | saxa | ok, so when i call from a to b the call enters the dialplan at what is written in contex= at boxb iax.conf |
| 12:35.42 | kaldemar | yes |
| 12:36.08 | bobg | i have an IAX trunk between an old 1.2.4 asterisk box and and new 1.6.2. Calls from 1.6 to 1.2 work fine but calls from the 1.2 to 1.6 fail with "all circuits are busy now" |
| 12:36.09 | zkn | to interconnect box a and box b using iax2, you don't necessarily need to use register => |
| 12:36.10 | zkn | what you can also do is set up a user and peer contexts separately on both box A and box B, define authentification method with a password for example, and deny any other connection but the permitted IPs |
| 12:37.10 | saxa | zkn: oh, ok, same as for a client |
| 12:37.17 | saxa | zkn: thx, i got it |
| 12:37.24 | zkn | maybe this approach is more confusing, but that's what I've been using |
| 12:37.44 | zkn | seems more secure |
| 12:37.55 | zkn | or not |
| 12:38.02 | zkn | not really a security guru |
| 12:38.26 | saxa | depends, the problem is that i'm getting some warnings and i see the boxb registers to boxa but then when i try to call i get everybody is busy |
| 12:39.39 | zkn | and what happens on the other box's console at the same time ? |
| 12:41.44 | saxa | http://pastebin.com/XXa6LwdF |
| 12:42.18 | saxa | zkn: on the other side i see nothing when I call from boxb to boxa |
| 12:42.28 | zkn | are you using IAX or IAX2 ? |
| 12:42.40 | zkn | i see: Unable to create channel of type 'IAX' (cause 66 - Channel not implemented) |
| 12:43.23 | saxa | does the file need to be named iax.conf or iax2.conf ? |
| 12:43.32 | zkn | maybe "IAX/brastrak" should be IAX2/brastrak |
| 12:43.45 | saxa | oh ok, need to check that |
| 12:43.55 | saxa | i think i used IAX2 |
| 12:44.09 | saxa | Name/Username Host Mask Port Status |
| 12:44.12 | saxa | brastrak/brastr 189.105.92.184 (D) 255.255.255.255 4569 (T) OK (429 ms) |
| 12:44.15 | saxa | 1 iax2 peers [1 online, 0 offline, 0 unmonitored] |
| 12:44.19 | saxa | thats what i see on boxa |
| 12:44.24 | zkn | check your dialplan |
| 12:44.25 | saxa | the one in Italy |
| 12:44.35 | zkn | Dial("DAHDI/1-1", "DAHDI/3&SIP/sasa&IAX/brastrak/1005,30") |
| 12:44.44 | kaldemar | saxa: "No channel type registered for 'IAX'" <--- replace IAX with IAX2 |
| 12:44.53 | saxa | so boxb registers ok to italy boxa |
| 12:44.57 | kaldemar | oh, that was found already. |
| 12:45.10 | saxa | ok let me check that |
| 12:45.48 | *** join/#asterisk ssureshot (~digitolx@12.196.90.82) |
| 12:46.11 | zkn | i'm sure that's your problems, which explains why on the other side you don't see any inbound calls from the other box |
| 12:46.17 | zkn | -s |
| 12:48.19 | saxa | http://pastebin.com/zUGsA7Y9 <-- thats still on box a |
| 12:48.42 | saxa | but now iax2 show peers on boths shows as registered |
| 12:49.32 | saxa | yes, i had on boxa (italy) the IAX/brastrak instead of IAX2/brastrak |
| 12:52.21 | ssureshot | I'm looking for some info,,, here is my question... I have two T1 lines with two asterisk servers.. A1 is primary, A2 is for failover, ie .. physically moving the server to primary location if the primary fails or.. implementing a CLAR service is the primary line goes down.. |
| 12:53.02 | ssureshot | Is there a failover solution that I could implement where I dont' need to physicaly move servers? |
| 12:53.38 | ssureshot | can I have the secondary server forward incomming calls to the primary if the primary T1 fails? |
| 12:54.26 | n3hxs | Arrange with Telco to forward to secondary if primary fails. |
| 12:55.21 | ssureshot | n3hxs: we do have that setup as a CLAR.. with emergency plans... |
| 12:55.29 | n3hxs | I am sure you can have server 2 direct DID to server 1 when server 1's T1 fails. |
| 12:56.29 | n3hxs | Though I have seen in the channel those who have set up that type of routing, I have not done it as I only have one server... and no T1 ;) |
| 12:56.30 | *** join/#asterisk imox1234 (~imox1234@p4FC5C763.dip0.t-ipconnect.de) |
| 12:57.19 | n3hxs | If you fill up all your talk paths on server 1 with the primary T1, then do they roll to the backup T1? |
| 12:57.23 | n3hxs | in which case they should route to the primary server. |
| 12:57.34 | ssureshot | n3hxs: that is what I'm looking for.. say I implement that emergency plan and they redirect the primary line to our backline... can that be directed to our secretary and she can then dispatch to the extensions |
| 12:57.55 | ssureshot | I do not have any rollover setup between the two servers |
| 12:57.58 | *** join/#asterisk JonnyD_work (~Jon@173.226.80.154) |
| 12:58.03 | ssureshot | they are completely seperate entities |
| 12:58.30 | saxa | [Mar 23 14:16:54] NOTICE[5288]: chan_iax2.c:11225 socket_process: Registration of 'rc-italy' rejected: '<unknown>' from: '189.105.92.184' |
| 12:58.44 | saxa | zkn: any idea why do i get this ? |
| 12:59.10 | n3hxs | ssureshot, do the calls come into the two separate T1s with different DIDs? |
| 12:59.13 | *** join/#asterisk Sheeplet (~multivac@41-133-81-114.dsl.mweb.co.za) |
| 12:59.25 | Sheeplet | lo all |
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| 12:59.57 | ssureshot | n3hxs: correct,,, they are completely seperate T1's with seperate DIDs |
| 13:00.09 | zkn | saxa. you mean chan_iax2.c:8621 reg_source_db: IAX/Registry astdb host:port invalid - '189.105.92.184:4569' ? |
| 13:00.29 | ssureshot | they even come in through seperate towns lol |
| 13:00.49 | saxa | zkn: no the NOTICE[5288] |
| 13:01.16 | saxa | [Mar 23 14:16:54] NOTICE[5288]: chan_iax2.c:11225 socket_process: Registration of 'rc-italy' rejected: '<unknown>' from: '189.105.92.184' |
| 13:01.35 | saxa | that rejection ^^ |
| 13:02.32 | zkn | any information on the box that rejected? |
| 13:02.34 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 13:03.08 | n3hxs | ssureshot, well, I was asking if there is a way for asterisk to differentiate between calls arriving that belong to that server and those which have failed over from the dead T1 issue. |
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| 13:04.21 | WIMPy | n3hxs: The called extension? |
| 13:04.39 | ssureshot | n3hxs: no there is not,, that is what somewhat what I want to accomplish.. |
| 13:05.12 | n3hxs | Yep, so calls hitting either of the boxes don't have DID info? Just a new call arriving. |
| 13:05.44 | *** join/#asterisk m_tadeu (~quassel@static-b5-252-50.telepac.pt) |
| 13:05.48 | saxa | zkn: no |
| 13:06.14 | n3hxs | ssureshot, You will need to get CLAR... to add DID so you can build routes to direct calls to the proper server. |
| 13:07.21 | saxa | zkn: http://pastebin.com/EhcDM2uh |
| 13:08.07 | ssureshot | n3hxs: I have the DID's setup in extensions.conf and can dial in using the DID's but all the phones would be registered to the primary asterisk server... I woud have to change change configs and services to get the phones to register to the backup server |
| 13:08.08 | zkn | don't see anything wring there |
| 13:08.14 | zkn | wrong |
| 13:08.15 | ssureshot | I have CLAR setup.. |
| 13:08.29 | saxa | zkn: yeah, to me also seems ok |
| 13:09.53 | zkn | saxa: you sure both server are using port 4569 and it is open in the firewall? |
| 13:10.11 | juliocesarlhg | help with mgcp.conf |
| 13:10.34 | zkn | saxa: this seems odd, still:IAX/Registry astdb host:port invalid - '189.105.92.184:4569' |
| 13:10.39 | n3hxs | ssureshot, BRB got an emergency to handle. Anyone else care to step in? |
| 13:11.35 | ssureshot | n3hxs: np man thanks |
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| 13:13.19 | saxa | zkn: i'm rechecking all things again |
| 13:13.43 | saxa | zkn: i do not get anymore that IAX/Registry error |
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| 13:14.46 | zkn | saxa: ok, what about calls, is it working now? |
| 13:16.20 | saxa | no |
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| 13:19.55 | m_tadeu | hi, I started to use wireshark to check a problem with RTP, but now I notice that one peer is getting packets with a bad checksum...has anyone went through this before? |
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| 13:47.53 | retentiveboy | Is there support in the Jabber features for joining on startup? |
| 13:48.13 | retentiveboy | Joining an XMPP chat room that is. |
| 13:52.12 | retentiveboy | Alternatively, is there a way to execute a portion of the dialplan on startup? Guess I could hack something together with a call file created in the start script. |
| 13:54.52 | saxa | zkn: i need to figure out why my 4569 port is not opened. Thx for now, need to go. |
| 13:54.53 | blitzrage | retentiveboy: use #exec |
| 13:55.25 | retentiveboy | blitzrage: new trick for me. will go do some reading. thx |
| 13:55.26 | blitzrage | extensions.conf >> #exec /path/to/script-stdout-goes-to-dialplan.sh |
| 13:55.44 | blitzrage | will be run each time a 'dialplan reload' is executed |
| 13:56.07 | blitzrage | can be used in an .conf file (except a couple, like asterisk.conf, etc) |
| 13:56.25 | retentiveboy | looks like it's for generating the config, right? |
| 13:56.26 | blitzrage | I use it in sip.conf a bunch to generate sip peers from a database if I don't want to use realtime |
| 13:56.52 | blitzrage | yes, but you can execute anything inside the script. Just don't echo anything to STDOUT if you don't want anything put into the dialplan |
| 13:57.12 | retentiveboy | I'm looking for a way to run JabberJoin on startup, not generate my dialplan |
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| 13:59.29 | tzafrir_laptop | blitzrage, hmm.. that does not execute dialplan |
| 13:59.33 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 13:59.47 | blitzrage | tzafrir_laptop: no, but you can easily connect to the AMI from the #exec script to trigger dialplan |
| 13:59.48 | tzafrir_laptop | It gets executed before the dialplan is read |
| 13:59.57 | blitzrage | ah, well that makes sense |
| 14:00.02 | tzafrir_laptop | So you can't assume your dialplan is in place |
| 14:00.09 | blitzrage | so use a callfile to execute dialplan a few seconds after the script is exectured |
| 14:01.07 | blitzrage | asterisk -rx "core waitfullybooted" to check when it is fully booted |
| 14:01.22 | blitzrage | there are certainly ways to make it happen |
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| 14:03.05 | blitzrage | or use cli.conf to automatically execute commands at startup |
| 14:03.35 | blitzrage | you can then use 'channel originate' to execute a Local channel to login using JabberJoin |
| 14:04.00 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:04.24 | blitzrage | I think that was 3 separate solutions, so take your pick :) |
| 14:06.17 | retentiveboy | That last one has me curious. Will poke around. Much thanks. |
| 14:07.14 | blitzrage | :) |
| 14:09.41 | jacc0 | if anybody is interseted in the solution a made for runing a AGI script async: http://pastie.org/1704021 |
| 14:10.02 | jacc0 | it look stupid, I know |
| 14:10.04 | jacc0 | :p |
| 14:10.34 | jacc0 | a=I |
| 14:13.39 | m_tadeu | hi, I started to use wireshark to check a problem with RTP, but now I notice that one peer is getting packets with a bad checksum...has anyone went through this before? |
| 14:14.32 | retentiveboy | Wireshark will report checksum errors when it's using a hardware offload engine on your Ethernet card. |
| 14:14.47 | retentiveboy | Probably nothing to worry about. Probably... |
| 14:15.04 | m_tadeu | ah I see... |
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| 14:19.34 | zkn | what causes this ? WARNING[21715]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 81.20.144.33 on IAX2/zeppelink0-7178 (type = 6, subclass = 11, ts=1209808, seqno=123) |
| 14:20.43 | blitzrage | a response was no received from the far end, so asterisk gave up |
| 14:20.52 | zkn | what happens there is the call is not hungup on the server even though it was hung up but he inbound caller |
| 14:21.10 | zkn | but= by |
| 14:21.39 | zkn | so on the soft phone the call remains and CLI shows me these errors |
| 14:22.56 | zkn | response was not received.. ok, makes sense if the caller hungup the phone...who does Asterisk fail to understand that the call was hung up? |
| 14:23.20 | zkn | this happens sporadically |
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| 14:29.04 | *** join/#asterisk hexanol (~etienne@modemcable094.94-70-69.static.videotron.ca) |
| 14:31.44 | hexanol | in chan_sip, is TCP and TLS support is still considered experimental or are the comments in the source code out of date ? |
| 14:32.16 | WIMPy | hexanol: Well, at least with TLS you can make Asterisk crash. |
| 14:34.09 | hexanol | I also got a crash while testing, but that might be only a bug, i.e. there's a difference between something crashing and something considered experimental |
| 14:35.23 | WIMPy | I don't think it's considered experimental in 1.8, but I do :-) |
| 14:35.50 | hexanol | ok |
| 14:36.02 | WIMPy | I think some random disconnects I had were liked to TLS usage as well. |
| 14:36.25 | WIMPy | But as they were pretty random, I don't really know. |
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| 14:45.27 | drift- | :) |
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| 14:47.37 | retentiveboy | blitzrage: I've got the dialplan setup for some startup logic and it works when I enter "channel originate ..." in the CLI but adding the same to cli.conf isn't working. Getting a warning for that line of cli.conf saying it's missing an equal sign. What should the line in cli.conf look like? |
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| 15:10.29 | upp | i'm looking for some one on europe know sipgate, because i have a trunk from sipgate but i don't get it online |
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| 15:22.55 | m_tadeu | I have RTP packets moving around when a client calls another client directly. Now if a client gets to the IVR first, goes into a queue, and only then the call is answered, I don't have RTP packets anymore |
| 15:24.22 | m_tadeu | both clients are using sip phones, and the one that iniciates the call is outside the asterisk nat, the one that gets the call is inside |
| 15:28.22 | psilikon | I am not clear on if it is absolutely necessary to forward port 5060-5080udp and 10000-20000 to an asterisk box behind a firewall. Seems that with pfsense I don't need to. My sip provider works great and I have two way audio but on a Verizon fios router I must forward all the ports right to the asterisk box and it gets attcked like crazy. Even with fail2ban it is annoying. |
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| 15:30.19 | *** join/#asterisk MRH2 (~chatzilla@62.49.242.3) |
| 15:32.48 | MRH2 | hi can I confirm digium ISDN hardware (T411P) provides a timing source even when there are no connected channels / not connected to an ISDN line. |
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| 15:41.50 | Naikrovek | MRH2: I think there's a dahdi_test command or something |
| 15:42.13 | MRH2 | <--is still talking zaptel lol |
| 15:42.46 | Naikrovek | well see if there's a zap_test command or something like that |
| 15:43.50 | Naikrovek | or you can just dial a meetme conference |
| 15:44.02 | Letoric | does anybody know, if including globals defined in other files has been removed? |
| 15:44.03 | Naikrovek | if there's no timing source, at least in my experience, meetmes won't work |
| 15:44.25 | Letoric | it's not working for me, and before I debug too much, figured I would ask if it was taken out since the info on voip-info.org is pretty dated |
| 15:44.49 | MRH2 | yank the cable and see if it works then - great lol |
| 15:48.47 | jacc0 | psilikon: For that reason we stated to use the geoip module for iptables and only allow ip's from countries we are doing busines with |
| 16:00.07 | Naikrovek | anyone in her eknow anything about peoplesoft? |
| 16:00.19 | Naikrovek | i realize this is #asterisk, i'm just asking |
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| 16:06.02 | AlHafoudh | hi all! |
| 16:07.17 | Naikrovek | all says "hi" |
| 16:07.26 | Naikrovek | hi |
| 16:07.29 | AlHafoudh | i am using openh323 (not ooh323) in my asterisk installation in order to work with avaya pbx and I am now trying to get TOS to work. I have set tos=lowdelay in h323.conf, but when I capture packets, the DSCP field is still 0x00, can someone help me? does openh323 library take care of the RTP stream TOS or generally asterisk is doing that? |
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| 16:15.22 | iamaham | greetings |
| 16:15.34 | iamaham | is there anyway to setup a sip phone so that it can answer other extensions calls? |
| 16:15.48 | iamaham | know it sounds silly but have to at least see if it's possible |
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| 16:22.55 | MRH2 | http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups |
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| 16:26.08 | iamaham | hrm so possible but almost no info on really how to do it |
| 16:26.19 | iamaham | ty googling now |
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| 16:28.59 | iamaham | hrm I dont have any user specific stuff in users.conf is that abormal |
| 16:29.10 | iamaham | all of that is in extension.conf and I think maybe sip.conf |
| 16:29.29 | iamaham | I see in users.conf i haev a callgroup 1 and pickupgroup =1 |
| 16:29.43 | iamaham | how do I set specific phones to which pickup group |
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| 16:31.15 | iamaham | back sorry |
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| 16:34.09 | iamaham | er another question say I chance features.conf can I reload that conf in a live system or do I have to restart it. (hard to do during the day) |
| 16:37.47 | MRH2 | ne1 know if there was any development of a controlplayback with multiple forward and rewind for 1.8 (closest I can remember was a controlplayback2 like https://issues.asterisk.org/view.php?id=8213) |
| 16:42.42 | iamaham | how do you add people to a callgroup or pickupgroup? I have *8 enabled in features, and the other variables are set to 1 in users.conf |
| 16:50.01 | iamaham | man there so many different versions of asterisk. trying to google info and I keep coming up with variations like tribox, freepbx, etc. hard to find info on just asterisk |
| 16:50.30 | iamaham | or using some gui I have no idea about. just want info on what .conf files to edit |
| 16:50.32 | iamaham | lol |
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| 16:54.38 | p3nguin | iamaham: You shouldn't be using users.conf. |
| 16:57.03 | p3nguin | iamaham: There are currently three active branches of Asterisk: 1.8, 1.6.2, and 1.4. Each of those branches only has one current version. Basically you have only three version to worry with -- that's not so many. |
| 16:58.07 | p3nguin | iamaham: To configure callgroups and pickupgroups so you can "hijack" a call to another phone, open up sip.conf and find the phone's peer definition... |
| 16:59.04 | p3nguin | iamaham: Decide on an arbitrary callgroup number for the phone to be in. If you chose 1, add "callgroup=1" to the phone's peer definition. |
| 16:59.21 | iamaham | woot worked had to add it on each persons entry in sip.conf and *8# works now |
| 16:59.39 | p3nguin | Weird. |
| 16:59.49 | AlHafoudh | anyone please? |
| 17:00.42 | p3nguin | alhafoudh: You may need to repeat the question; I didn't see it. |
| 17:01.31 | elb | p3nguin: as a point of interest, my unreliable gtalk incoming line appears to have become reliable some time yesterday -- with no changes to configuration |
| 17:01.45 | elb | p3nguin: in addition, while I previously did not have to press 1 to accept the incoming call, I now do |
| 17:02.24 | p3nguin | elb: If you have to press 1 to accept, it is working correctly. |
| 17:02.45 | p3nguin | elb: I configure dial plan to accept the call for me. |
| 17:02.59 | elb | yeah, I did, too |
| 17:03.07 | elb | and I know it's supposed to require that ... but it didn't |
| 17:03.09 | elb | and now it does |
| 17:03.20 | elb | (as well as work reliably) |
| 17:03.28 | elb | I dunno, maybe Google had something going on |
| 17:04.33 | p3nguin | Maybe it's possible, but no one else complained about any problems during the same time frame. |
| 17:05.24 | elb | that's not really relevant with modern services |
| 17:05.34 | elb | I'm a Pidgin developer ... we *often* see isolated server errors |
| 17:05.41 | elb | or errors affecting only a small number of users |
| 17:05.55 | elb | since services are now spread across a zillion machines in dozens of data centers, stuff happens |
| 17:06.09 | *** join/#asterisk msetim (~setim@186.214.197.221) |
| 17:06.11 | msetim | Hi guys |
| 17:06.28 | elb | anyway, I didn't touch a configuration file or restart or reconfigure asterisk in 24h ... and it went from unreliable with the Google servers reporting internal server errors on 1/3 calls, to working every time |
| 17:10.37 | Freeaqingme_ | gratz! |
| 17:10.49 | Freeaqingme_ | I suppose google is also finetuning some stuff here and there |
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| 17:37.54 | malcolmd | elb: definitely been a lot of weirdness w/ google voice lately. i always had to press 1, but i don't have to do a wait. others need a long wait (8 seconds) and/or have to send 1 twice |
| 17:39.07 | Naikrovek | are you guys tying asterisk into GV? is there a sanctioned way to do that now |
| 17:39.33 | malcolmd | xmpp is the way that's worked for years now |
| 17:39.39 | Naikrovek | <-- been out of the world for a spell |
| 17:39.55 | Naikrovek | ah so same old hack. i read recently that they were rolling out a sip service |
| 17:39.56 | malcolmd | 1.6 tied into old google chat client (windows only). 1.8 ties into google chat (web client) and google voice |
| 17:40.27 | Qwell | That's a little misleading. It doesn't actually use the web client at all. Just the same service it does. |
| 17:40.32 | malcolmd | the sip service doesn't seem to be sanctioned currently, it's been pulled a couple of times |
| 17:40.40 | Qwell | it's all native though |
| 17:41.32 | malcolmd | Qwell: apologies. by web client i meant that calls can be placed from google chat web clients and asterisk and vice versa, provided you're using 1.8. with 1.6 you could only call people using google chat applications, not the web client. |
| 17:41.43 | malcolmd | more of a functional description than a process description |
| 17:42.12 | Qwell | huh. 1.6 couldn't call people using the web client? |
| 17:42.19 | malcolmd | nope. |
| 17:42.26 | Freeaqingme_ | Naikrovek, given the amount of questions popping up everywhere I'd hold off to that for now |
| 17:42.26 | Qwell | rephrase for disambiguation: |
| 17:42.29 | Qwell | huh. 1.6 couldn't call people that were using the web client? |
| 17:42.35 | malcolmd | nope |
| 17:42.40 | Qwell | weird |
| 17:42.42 | malcolmd | yup |
| 17:42.48 | Naikrovek | not to worry, i have no intention of doing it, i was just curious if there was a sancitoned way to do it now |
| 17:42.52 | Qwell | though I guess the web service isn't gtalk |
| 17:44.22 | malcolmd | i think it's officially "google chat" |
| 17:44.45 | malcolmd | and my definition of "official" means "that's what google calls it in your google voice settings" ;) |
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| 17:45.59 | m_tadeu | I don't have rtp packets moving around when the call is answered, after the caller waits in a queue. what can be the problem? both are sip, the caller outside nat and the callee inside |
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| 17:50.27 | blitzrage | has a personal pet peeve about referring to any Asterisk version as "1.6" |
| 17:50.36 | blitzrage | since no such branch exists |
| 17:50.52 | blitzrage | and feature set of 1.6.0/1.6.1 != 1.6.2 |
| 17:52.01 | Freeaqingme_ | m_tadeu, what vresion are you running? |
| 17:53.27 | brainiac | I have a question about how Voicemail() is supposed to behave with a full mailbox. Will it still send email if the mailbox is full? |
| 17:53.45 | blitzrage | I'd not expect it to since no new voicemail should be able to be recorded |
| 17:53.53 | blitzrage | you'd just get a prompt saying the mailbox is full |
| 17:54.56 | brainiac | that's what I thought. How is a 'full' box defined... # of msgs or amt of available storage? |
| 18:00.10 | blitzrage | brainiac: see 'maxmsg' in voicemail.conf.sample |
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| 18:05.57 | DaveH2 | Hi all |
| 18:06.25 | DaveH2 | Is this the right place to ask about an asterisknow installation? or is there a dedicated channel for that? |
| 18:07.14 | nestAr | yeah, #asterisknow IIRC |
| 18:08.57 | DaveH2 | excellent thanks :) |
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| 18:16.20 | DaveH2 | as my actual issue is asterisk based (rather than the gui), I hope it will be ok for me to cross-post/ask in here as well... I'm running asterisknow 1.7.1 with freepbx, and I have 2 extension and a 3 port tdm410 fxo card.. I can make incoming/outgoing calls ok.. but there is no caller id appearing on my phones (snom m3). |
| 18:16.31 | DaveH2 | before asterisknow.. I was running the free edition of switchvox and caller id worked ok, so I'm assuming it's a config I'm missing out on, or is there a known issue with asterisk 1.6.x and it's worth me upgrading to 1.8? |
| 18:17.08 | nestAr | no, CID should work fine in 1.6.x |
| 18:17.53 | nestAr | i'm assuming CID works from phone to phone, just not on incoming calls over the TDM? |
| 18:20.41 | DaveH2 | yup, that is correct |
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| 18:35.37 | jmchado | anyone here done any work installing Asterisk on a mobile phone? |
| 18:36.10 | Freeaqingme_ | jmchado, I havent. But, why on earth would you want to do so? |
| 18:36.20 | jmchado | I have flashed the ROM on my cell and installed Linux OS (was using windows mobile 6.5 |
| 18:36.32 | *** join/#asterisk AMindMobile (~AMindMobi@95-28-106-82.broadband.corbina.ru) |
| 18:36.36 | jmchado | well I want to make my phone my PBX |
| 18:36.48 | *** join/#asterisk gruvfunk (4c0f2988@gateway/web/freenode/ip.76.15.41.136) |
| 18:37.39 | jmchado | I believe I can create a truly mobile VoIP trunked GSM service |
| 18:38.03 | jmchado | I have an USRP running OpenBTS |
| 18:39.19 | gruvfunk | How can I pause/hold/park a caller in an IVR, generate some outbound calls to collect information, and feed that data back to my original caller? |
| 18:39.19 | jmchado | and I'm thinking I can host hte PBX on my mobile which already controls the USRP |
| 18:42.09 | *** join/#asterisk hfb (~hfb@pool-96-247-49-75.lsanca.dsl-w.verizon.net) |
| 18:42.12 | blitzrage | gruvfunk: use an AGI() |
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| 18:47.29 | jmchado | I am trying to realize a cost effective solution for NPOs who aid in disaster relief, by setting up highly mobile pico cells to increase the cellular coverage in areas were usage goes up 10x (i.e. Japan atm) due to the amount of aid workers and NPOs occupying previously rural areas |
| 18:48.15 | jmchado | I have the two parts disassociated, and want to see if anyone has done something similar to make them compatible |
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| 18:48.33 | *** join/#asterisk gruvfunk (~paul@user-160uac8.cable.mindspring.com) |
| 18:48.57 | jmchado | I have a phone running Debian hosting OpenBTS on a USRP, and would like to trunk my little pico cell to a VoIP provider |
| 18:49.03 | MRH2 | wouldn't u want asterisk as part of or next to the USRP? you can always remote in to control it? |
| 18:49.30 | jmchado | and use the same hardware (mobile with OS) as the asterisk server |
| 18:50.04 | jmchado | So eventually I am going to host multiple USRPs with the same server |
| 18:50.23 | gruvfunk | IVR HELP -> How can I park/pause a caller in IVR, fire off some outbound calls to collect information and then present that data back to my original caller? Is this a Park, ConfBridge, what? |
| 18:50.53 | gruvfunk | Context Jumping? |
| 18:51.03 | Freeaqingme_ | gruvfunk, as blitzrage mentioned already, use AGI for that |
| 18:51.26 | MRH2 | is that the same sort of thing as http://www.tombom.co.uk/blog/?p=144 |
| 18:51.49 | gruvfunk | sorry, missed that.. my irc client crashed - was there any other info (repaste, please) |
| 18:52.11 | jmchado | MRH2 checking now |
| 18:53.12 | Freeaqingme_ | gruvfunk, nope |
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| 19:11.32 | gruvfunk | I am in need of some IVR + AGI assistance if anyone has a couple of minutes to give me an overview and point me in the right direction. PM me pls, thx! |
| 19:11.48 | brightidea1980 | any solution or workaround to high CPU usage in asterisk 1.8.3.2 - I have tested the same in centos linux box and in a mac osx 10.6.6 box and problem persists |
| 19:12.47 | Letoric | Anybody able to suggest a method for me to check if a sip peer is reachable prior to sending a call to it? |
| 19:13.09 | Letoric | I have them qualified, so I have the state, just not sure how to query it yet |
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| 19:14.41 | *** join/#asterisk Docfxit (~none@netblock-75-79-6-149.dslextreme.com) |
| 19:16.41 | Docfxit | When I enter sudo asterisk -vrrrrrr I get an error saying: Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) The file does exist with zero bytes. Is there supposed to be something in the file? |
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| 19:17.37 | psilikon | Docfxit, you sure asterisk is running "ps fax | grep asterisk" |
| 19:19.10 | zkn | brightidea1980: have you tried 1.8.3 ? is 1.8.3.2 causing more problems? |
| 19:19.25 | psilikon | Docfxit, maybe I should have phrased it like this: Are you sure that asterisk is running? You can verify that it is running with the following command, "ps fax | grep asterisk". |
| 19:20.31 | elb | malcolmd: interesting that you say it can call the web client ... I've been unable to make that work |
| 19:20.34 | psilikon | Letoric, what about this: http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanIsAvail |
| 19:20.40 | elb | asterisk says there's nobody to call, or some such |
| 19:21.06 | elb | (buddy list entries are reciprocal, so I don't think it's a blocking problem) |
| 19:21.41 | psilikon | Letoric, I am not sure if that will work but that is the first result google returned from "check if sip peer is reachable" |
| 19:23.34 | brightidea1980 | member:zkn actually I have tried 1.8.3.1 which had the same problem - I have also googled for similar user reports and found this https://issues.asterisk.org/view.php?id=18569 |
| 19:23.53 | Docfxit | psilikon What I get back from that command is three lines. Would you like me to post it here? |
| 19:24.31 | psilikon | Docfxit, probably be best to just paste it somewhere. We don't wanna annoy ppl here. |
| 19:24.33 | brightidea1980 | it seems to be a long standing issue in 1.8 branch of asterisk |
| 19:25.38 | benngard | Letoric: core show function DEVICE_STATE |
| 19:26.43 | Docfxit | psilikon You can see the results at: http://pastebin.com/Z4Hc4p6D |
| 19:27.40 | zkn | <brightidea1980: i wonder which of the 1.8 versions is the most stable |
| 19:28.46 | brightidea1980 | that's exactly what's going on my mind - I don't want to lose the better support for gtalk in 1.8 which is not offered in earlier branches but I don't know which version to switch to |
| 19:29.36 | benngard | i am running 1.8.4-rc2, works very well for me |
| 19:30.12 | psilikon | Docfxit, Maybe because of the -f causing it to not fork. Kinda just a guess though. |
| 19:30.14 | brightidea1980 | I am using 1.8 @ home and 1.6 @ work, I couldn't upgrade because of stability issues in 1.8 |
| 19:30.20 | *** join/#asterisk jkroon (~jkroon@dsl-241-231-146.telkomadsl.co.za) |
| 19:30.21 | benngard | but with that many ooh323 patches i dont know if i can call it 1.8.4-rc2 anymore :) |
| 19:31.07 | Docfxit | psilikon Could it be the properties of the file. |
| 19:31.13 | psilikon | Docfxit, or because -c is outputting to the console. |
| 19:31.33 | psilikon | Docfxit, how was asterisk started? |
| 19:31.45 | Docfxit | psilikon The owner is root with only the owner having writes to write to it. |
| 19:32.08 | Qwell | elb: Are you using Asterisk from a package? |
| 19:32.21 | Docfxit | psilikon It starts auto when the machine starts up. |
| 19:32.36 | psilikon | Docfxit, what distro? How did you install *? |
| 19:32.59 | benngard | debian, ubunt, gentoo... ? |
| 19:33.19 | psilikon | Docfxit, I bet if you did a ctrl+alt+F[2-6] you would see asterisk. |
| 19:33.47 | psilikon | maybe ctrl+alt+F1 too I guess. |
| 19:34.31 | Docfxit | psilikon Where would I press the keys? From terminal? |
| 19:35.34 | psilikon | Docfxit, yeah. Describe you system and asterisk install method please. Are you physically in front of this asterisk machine? What distribution of linux is it? How did Asterisk get installed? |
| 19:37.43 | Docfxit | psilikon I'd be happy to give you all the info. I now have another problem. After pressing ctrl alt f2 I have many bars across the screen so I can't see any windows. |
| 19:38.30 | gruvfunk | Doc, slow down and answer questions |
| 19:39.02 | Docfxit | psilikon I have the screen back now. I'll answer your questions. |
| 19:39.09 | psilikon | Docfxit, hehe ok. |
| 19:40.44 | Docfxit | psilikon It's installed in Ubuntu , |
| 19:41.09 | psilikon | What version you running? |
| 19:41.32 | psilikon | cat /etc/issue |
| 19:42.01 | Docfxit | psilikon I did get the windows back but I can't enter anything into the screen now. |
| 19:42.03 | psilikon | lsb_release -a might work too |
| 19:42.30 | Qwell | Why are you running X on a server with Asterisk? |
| 19:42.36 | Qwell | Step 1: Don't do that. |
| 19:43.07 | Docfxit | psilikon I'm not in front of the machine. I'm remote through VNC. |
| 19:43.19 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 19:43.24 | psilikon | Docfxit, handy tidbit of info to share ;) |
| 19:43.26 | Qwell | blinks |
| 19:44.34 | psilikon | Docfxit, I would love to stay and help out but I gotta hit the road. |
| 19:44.53 | psilikon | Docfxit, when in doubt just reboot. |
| 19:44.59 | Docfxit | psilikon Thank you very much for your help. |
| 19:45.20 | Docfxit | psilikon I'll keep trying to get control back. |
| 19:45.32 | Docfxit | psilikon Have a good day. |
| 19:45.37 | psilikon | Docfxit, np. |
| 19:45.42 | psilikon | have a good one |
| 19:45.53 | Docfxit | psilikon Thanks. |
| 19:50.03 | Letoric | psilikon: I ended up going with function_SIPPEER since I have it qualified. Now I just have to overcome the challenge of making GotoIf process a word comparison, vs number comparisons. Still a newb ;) |
| 19:50.20 | benngard | i need to know the status of a cell phone so i did like this (wonder if it is totally wrong way to do it?): i use the callforward feature of the cell phones, i use number "a" for busy and "b" for unavailable, dial the mobile for a second and set status = not_in_use, if the cell phone is busy i get a call back to "a" use that call for setting status = busy and ofc if a get a call back to "b" i set status = unavailable, it works, but |
| 19:52.33 | Docfxit | Letoric Please note psilikon had to leave for now. |
| 19:53.38 | Letoric | oh, sorry ;) |
| 19:54.15 | Letoric | well can anybody provide some guidance for me on this? I've tried quotes and still it's always going to the 'success' match for the gotoif statement |
| 19:54.26 | Letoric | trying to compare for UNREACHABLE |
| 19:54.47 | Letoric | if it matches, go to success section, else go to failure section |
| 19:55.39 | Letoric | I have exten => PSTN1LOCALCHECK10,n,Gotoif("${TRUNK_STATUS}"="UNREACHABLE"?10:20) |
| 19:55.50 | Letoric | it always goes to 10 |
| 19:55.59 | MRH2 | trs-80 programming BASIC flashback |
| 19:56.23 | Letoric | it's been awhile since I programmed in the trs80 ;) |
| 19:56.26 | Letoric | I think I was 11 |
| 19:56.54 | elb | Qwell: negative, 1.8.3.2 built from source |
| 19:58.49 | MRH2 | yeah i remember having to record the files to cassette, no kid had the money for a floppy drive |
| 19:59.11 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 19:59.11 | Freeaqingme_ | gruvfunk, figure out your agi stuff? |
| 19:59.43 | gruvfunk | Freeaqigme, not yet |
| 20:00.03 | elb | Letoric: you're missing $[] |
| 20:00.08 | gruvfunk | pointers welcome |
| 20:00.22 | *** join/#asterisk jkroon (~jkroon@dsl-241-231-146.telkomadsl.co.za) |
| 20:00.36 | elb | Letoric: PSTN1LOCALCHECK10,n,GotoIf($[${TRUNK_STATUS}="UNREACHABLE"]?10:20) |
| 20:00.51 | zkn | why does some like that happen?? :( WARNING[21706] chan_sip.c: sip_xmit of 0x87ffc90 (len 586) to 192.168.3.211:53645 returned -2: Interrupted system call |
| 20:00.52 | blitzrage | elb: that won't work |
| 20:01.03 | blitzrage | elb: unless ${TRUNK_STATUS} is going to return with |
| 20:01.10 | blitzrage | " " around the result |
| 20:01.22 | blitzrage | otherwise you're goign to literally compare: FOO = "FOO" |
| 20:01.24 | elb | OK, that's fine, I don't know what's in that variable |
| 20:01.28 | elb | my point was, he's missing $[] |
| 20:01.44 | blitzrage | you'd need $["${TRUNK_STATUS}" = "UNREACHABLE"] |
| 20:01.49 | elb | the rest of his logical errors come later, if any ;-) |
| 20:01.54 | *** join/#asterisk UnixDev (~UnixDev@unaffiliated/unixdev) |
| 20:02.06 | blitzrage | there is no point in introducing additional syntax errors |
| 20:02.14 | elb | fair enough |
| 20:02.39 | elb | did I leave off quiotes he had in his original line? |
| 20:02.52 | elb | OK, yes, I did, apologies |
| 20:03.00 | blitzrage | I don't know what the original line was. I just sat down. |
| 20:03.25 | Letoric | thanks blitzrage, I had just worked it out and looked over to see you giving it to me haha |
| 20:03.37 | Letoric | you too elb ;) |
| 20:03.39 | Qwell | woah, it's blitzrage |
| 20:03.44 | blitzrage | heck ya it is |
| 20:06.40 | *** join/#asterisk aerecords (~IceChat77@static-87-102-95-10.karoo.KCOM.COM) |
| 20:07.20 | aerecords | hello all i need some installation help i am running on a ubuntu 9.04 vps and can't for the hell of it get asterisk realtime to work |
| 20:07.34 | aerecords | could somebody please help |
| 20:09.18 | PhreeBeer | can anyone anyone point a newb to some decent "overview" documentation? I'm trying to just organize things in my head while learning the whole telephony thing. Here's what I'm looking to do: replace my current company (proprietary) pbx with Asterisk, but I'm trying to figure out if I can keep my existing handsets. (also proprietary - Panasonic) |
| 20:09.54 | aerecords | are all the handsets the same |
| 20:10.10 | Naikrovek | how many handsets |
| 20:10.32 | PhreeBeer | same family. One is the "receptionist" handset with more functions. |
| 20:10.43 | aerecords | model numbers? |
| 20:10.47 | PhreeBeer | running about 7 handsets atm |
| 20:11.08 | PhreeBeer | KX-T7431 is the basic unit |
| 20:11.35 | elb | PhreeBeer: there are two books on Asterisk that I know of, both of which are freely available and both of which give a fair overview of what Asterisk can and can't work with |
| 20:11.49 | Qwell | ~book |
| 20:11.50 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 20:11.51 | Qwell | ~newbook |
| 20:11.51 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 20:12.04 | aerecords | have you done a basic google search to see if the handsets are compatible? |
| 20:12.16 | wdoekes2 | aerecords: "asterisk realtime" can mean lots of things.. I'm assuming realtime-dynamic sip-friends? |
| 20:12.22 | elb | there you go, I was getting URLs :-) |
| 20:12.32 | *** join/#asterisk lejocelynbis (~lejocelyn@nap13-6-88-179-18-139.fbx.proxad.net) |
| 20:12.41 | PhreeBeer | I'm having a hard time finding that info out - there's not a whole lot of info out there on them. (Old system) |
| 20:12.47 | aerecords | yes i want my staff to be able to logon to asterisk from data in my mysql database |
| 20:13.32 | wdoekes2 | logon? you mean some kind of agent logon? |
| 20:13.37 | PhreeBeer | I grabbed the Future Book. When I read through it some time ago, I didn't see anything specific to my situation. But I can have a harder look through again. |
| 20:13.47 | MRH2 | I miss the agent channel |
| 20:13.57 | elb | PhreeBeer: do you even know what *kind* of handsets those are? |
| 20:14.04 | *** join/#asterisk Freeaqingme_ (~dolf@dsl-083-247-011-232.solcon.nl) |
| 20:14.08 | lejocelynbis | hi, I'm trying to configure asterisk and asterisk-gui on a bsd system, when I try to connect on http://192.168.0.1:8088/asterisk/static-http/config/cfgbasic.html, the error message (not found) stil shows it's an asterisk server |
| 20:14.17 | elb | (pots, voip, proprietary digital, etc.) |
| 20:14.22 | lejocelynbis | I don't understand why it doesn't find the page, any ideas ? |
| 20:14.41 | PhreeBeer | The really basic literature says they're "digital" and, of course, work with the Panasonic pbx I wish to replace. |
| 20:14.52 | elb | uh oh ;-) |
| 20:14.56 | aerecords | yes so that when my staff opens the sip client it authenticates to the database and allows my staff to send and receive calls |
| 20:15.12 | *** join/#asterisk zkn (~zkn@82.131.54.59.cable.starman.ee) |
| 20:15.33 | elb | you'll probably need special hardware to interface those, if it can be done at all ... but I'm not the person to tell you about that, as I've never done it |
| 20:15.54 | PhreeBeer | I'm kind of getting that impression. :( |
| 20:15.54 | lejocelynbis | do we need to do a symbolic link to apache ? |
| 20:16.01 | blitzrage | those handsets are likely not useful without the panasonic system |
| 20:16.17 | wdoekes2 | aerecords: you don't *need* realtime for that.. you want realtime when you want to configure the users from a different system or if you have very many of them |
| 20:16.37 | blitzrage | any adapters for them will be just as expensive as an ATA or a new SIP phone |
| 20:16.42 | MRH2 | I'd run a softphone on the desktop in parallel with the 'live' panasonic system until you are comfortable |
| 20:16.56 | aerecords | it is for mass amounts of staff who work from a home basis |
| 20:16.56 | blitzrage | you'll be better off just selling the old system and phones and recoup some costs on the new phones and system |
| 20:17.04 | PhreeBeer | yeah.. that's what I wanted to figure out next :) |
| 20:17.25 | aerecords | phreebeer ive just done a quick google search on compatibility and apparently it is possible |
| 20:18.03 | PhreeBeer | you found something? cool. Maybe I was using the wrong terms (or to newbish to make the connection) |
| 20:18.53 | *** join/#asterisk blitzrage (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:18.53 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 20:19.04 | wdoekes2 | ok, the very many option. well.. it's not that difficult: (1) get the odbc (or other db) link to work, (2) config the extconfig for sippeers, (3) watch the sipfriend/sipreg table get populated with ipaddr/port info |
| 20:19.08 | aerecords | just looked to see if its a hybrid type phone system and it is just look for wither the terms hybrid or voip compatible |
| 20:19.55 | PhreeBeer | I'm pretty sure these phones pre-date voip. Got 'em in the days of dial-up. :P |
| 20:20.02 | blitzrage | yes they do |
| 20:20.09 | aerecords | i read somewhere that the users have to log in before the realtimes command will show peers |
| 20:20.23 | blitzrage | they will be analog phones (in which you can use an ATA) or they'll be digital sets keyed to the panasonic system specifically |
| 20:20.30 | wdoekes2 | you won't see any peers at all, unless you use rtcachefriends |
| 20:20.31 | blitzrage | aerecords, the peer has to register, yes |
| 20:20.39 | blitzrage | and that ^^^ |
| 20:20.45 | wdoekes2 | sip show peer 12345 load <-- without the load, you won't see them |
| 20:20.47 | aerecords | phreebeer visit http://www.docstoc.com/docs/25322294/Hybrid-IP-PBX-System-KX-TDA15KX-TDA30KX-TDA100KX-TDA200 |
| 20:21.21 | PhreeBeer | aerecords: thanks for the link! |
| 20:21.46 | *** join/#asterisk koffel (~whome@c-174-57-81-113.hsd1.pa.comcast.net) |
| 20:22.16 | *** join/#asterisk andygraybeal (~andy.gray@obsidian.casanueva.com) |
| 20:22.23 | PhreeBeer | meh. got to run. thanks for the help so far! |
| 20:22.23 | aerecords | welcome and ill try getting that to work now i can get it to connect to the database but the sip clients don't seem to want to register to it so ill try |
| 20:23.02 | aerecords | ciya Phreebeer |
| 20:23.05 | *** part/#asterisk PhreeBeer (~chatzilla@207.179.191.21) |
| 20:23.29 | wdoekes2 | well.. if the 'sip show peer ... load' shows nothing, steps 1/2 are bad, if it does show something, the registrations are bad.. but you should be able to trace that with 'sip set debug on' |
| 20:24.00 | aerecords | is that a cli command? |
| 20:24.19 | wdoekes2 | yes.. 'core show help sip set debug' |
| 20:24.59 | aerecords | ok ill try it now hopefully it will work if nt hopefully will talk later |
| 20:25.39 | wdoekes2 | good luck |
| 20:25.47 | aerecords | thanks |
| 20:26.19 | koffel | i have a question on amp files |
| 20:30.18 | *** join/#asterisk jkroon (~jkroon@dsl-241-231-146.telkomadsl.co.za) |
| 20:30.27 | koffel | now how do the amp files work with asterisk |
| 20:31.03 | Letoric | ok, another fun one, well, at least for me :P My Cisco call manager responds to multiple conditions, with the same generic 'Congestion' message. Is there a way for me to glean more from that, so that I can react appropriately to messages? |
| 20:32.05 | Letoric | I'm trying to write some fail-safe things in the dial plan, so it always tries our backup SIP provider when the router has issues, but sometimes the router 'issue' is that a person dialed 10 digits when it needed to be 11.....and I can adapt to that, but I don't know how to adapt when it responds with congestion because the PRI interface is down |
| 20:32.43 | Letoric | I can see that asterisk understands a difference in teh message, as the console shows 'Sip/pstn1-xxxxx is circuit-busy |
| 20:33.18 | *** join/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
| 20:34.51 | Get_The_Fish | Any one know of any plans to include a RHEL 6 repo for Asterisk? |
| 20:35.08 | Get_The_Fish | or should I just use epel? |
| 20:35.34 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:35.34 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:36.41 | Get_The_Fish | @leifmadsen do you know if a yum repo for RHEL 6 is in the cards any time soon? |
| 20:36.52 | leifmadsen | no idea, ask Qwell |
| 20:36.55 | Get_The_Fish | k |
| 20:37.00 | Get_The_Fish | thanks |
| 20:37.13 | Qwell | There will be some time after CentOS 6 gets released. |
| 20:37.23 | Get_The_Fish | ah, waiting on Centos. |
| 20:37.48 | Get_The_Fish | when is Centos going to be ready, you ask? When it's ready! |
| 20:37.55 | leifmadsen | it's not released? then why would there be RPMs for it now? |
| 20:38.06 | leifmadsen | that doesn't make any sense |
| 20:38.08 | Get_The_Fish | not that I know of |
| 20:38.27 | Get_The_Fish | nothing on the site |
| 20:38.57 | Get_The_Fish | I could just use epel, right? not that much different |
| 20:39.28 | *** join/#asterisk appel11 (~root@78-22-118-226.access.telenet.be) |
| 20:40.17 | appel11 | hello, got a bit of problem, I've got a asterisk pbx that got hacked, is it possible for a peer to create an extra peer? |
| 20:40.31 | appel11 | I've changed all the sip passwords |
| 20:40.37 | Get_The_Fish | not that I know of |
| 20:40.45 | appel11 | can't see him in the connected peers |
| 20:40.57 | appel11 | but still trying to make calls |
| 20:41.02 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 20:41.09 | appel11 | (stopped him with an authentica function |
| 20:41.38 | brightidea1980 | zkn: I found the problem in my asterisk 1.8.3.2 installation. Activating the pbx_spool.so will cause the process cpu usage to get stuck at around 100% most of the time; I have removed that module. I will monitor the system for a few more days to form a solid conclusion |
| 20:42.05 | zkn | what's the module for, do you know? |
| 20:43.11 | brightidea1980 | I am not sure, trying to find out right now |
| 20:44.26 | *** join/#asterisk aerecords (~IceChat77@static-87-102-95-10.karoo.KCOM.COM) |
| 20:45.02 | appel11 | is their a way that a client can offer a INVITE and asterisk will accept it even it isn't from an known peer? |
| 20:45.29 | aerecords | hello im back im stuck on where to put the relatime switch comment |
| 20:45.38 | aerecords | sorry realtime switch |
| 20:50.46 | zkn | brightidea1980: pbx_spoolOutgoing spool support relating to Asterisk call files |
| 20:51.54 | zkn | in my 1.8.3 installation i have put it on the noload list already |
| 20:52.04 | *** join/#asterisk personaljosh (~personalj@ip184-131-173-82.adsl2.static.versatel.nl) |
| 20:53.24 | *** join/#asterisk Get_The_Fish (~sbrady@c-24-8-50-199.hsd1.co.comcast.net) |
| 20:53.54 | personaljosh | hi all, i am having a problem with my asterisk installation running on OPENBSD latest version, the problem is i keep receiving the error message "Unable to change ownership of /var/run/asterisk/asterisk.ctl: Operation not permitted" .. when running safe_asterisk . --- if i do not run safe asterisk, and only asterisk, it loads fine but no sip drivers or iax drivers or to be honest any drivers are loaded.. i am happy to |
| 20:53.54 | personaljosh | pay for support and am possibly looking for a long term freelance technician at the same time.. thanks in advance |
| 20:54.11 | leifmadsen | go to OpenBSD then zoned out |
| 20:54.49 | personaljosh | leifmadsen , i have asked in both channels |
| 20:55.07 | leifmadsen | I'm just saying I got to OpenBSD then I stopped reading as I don't use OpenBSD |
| 20:55.31 | personaljosh | thankyou for your effort |
| 20:55.36 | brightidea1980 | zkn: I wasn't aware of the risk of not deactivating unneeded modules |
| 20:56.21 | brightidea1980 | zkn: anyway, I am going to be watching the server closely in the next few days to make sure nothing else causes freezing of the server |
| 20:57.04 | appel11 | hello, is it possible to create an account via the manager login? |
| 20:57.06 | zkn | i needed to restart Asterisk at least every day when I compiled it with all modules and kept the sample modules.conf unedited |
| 20:58.17 | appel11 | the thing is, i've got a server that being/is hacked atm and can't find the user in the configuration files |
| 20:58.21 | brightidea1980 | zkn: good advice - I am going to try to reduce the # of loaded modules |
| 20:58.57 | zkn | i used the Book as a reference what to disable and what not |
| 20:59.46 | *** join/#asterisk Get_The_Fish (~GTF@c-24-8-50-199.hsd1.co.comcast.net) |
| 20:59.56 | zkn | when I disabled the same modules in menuselect then to my surprose this removed some functions that it was not supposed to remove |
| 21:00.16 | personaljosh | guys i'm looking for an asterisk tech who is savvy with openbsd.. freelance work often needed i'm willing to pay $100/per hour |
| 21:00.16 | *** join/#asterisk jkroon (~jkroon@dsl-241-231-146.telkomadsl.co.za) |
| 21:01.20 | zkn | so reducing modules with modules.conf seems much better option |
| 21:01.20 | brightidea1980 | zkn: which book exactly was your reference? I haven't such a thing |
| 21:01.24 | zkn | the Boook :) |
| 21:01.35 | zkn | http://ofps.oreilly.com/titles/9780596517342/ |
| 21:01.38 | brightidea1980 | :) |
| 21:02.20 | brightidea1980 | zkn: nice .. so I have learned three things since I joined this channel |
| 21:04.34 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan.noare-1.holmedal.net) |
| 21:10.56 | *** part/#asterisk Get_The_Fish (~GTF@c-24-8-50-199.hsd1.co.comcast.net) |
| 21:11.13 | *** join/#asterisk Get_The_Fish (~GTF@c-24-8-50-199.hsd1.co.comcast.net) |
| 21:12.02 | brightidea1980 | thank you zkn, l8r guys. |
| 21:22.30 | appel11 | ok, somebody any advice? our asterisk server is somehow hacked, they are still trying to call from an sip user 1111 but that doesn't exist, how is that possible? |
| 21:22.56 | WIMPy | You let everyone in? |
| 21:23.20 | elb | are you sure you aren't just seeing someone trying to call extensions and failing? |
| 21:23.25 | elb | that happens a *lot* |
| 21:24.51 | *** join/#asterisk lordvadr (~something@jose-tc.ctc.biz) |
| 21:26.24 | appel11 | nope |
| 21:26.36 | appel11 | I send them to my phone for now |
| 21:26.58 | appel11 | lets say they speak with an arab accent |
| 21:27.19 | appel11 | how is it possible |
| 21:27.30 | appel11 | ow i missed a call there |
| 21:27.43 | appel11 | they are not very friendly |
| 21:28.13 | appel11 | they don't even greet me with their name :( |
| 21:29.33 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 21:30.32 | WIMPy | They most probably don't want to talk to you. |
| 21:30.46 | appel11 | perhaps ;) |
| 21:30.55 | appel11 | french this time |
| 21:31.07 | appel11 | I'm not good at that :( |
| 21:31.41 | gruvfunk | appel11 -> restrict your firewall to only allow traffic from your actual users? |
| 21:32.15 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 21:32.41 | appel11 | thing is, for now its not really a problem, it's my brothers pbx (his company) and there is nobody thier. But I cant figure out how they are still connected |
| 21:32.50 | appel11 | well its a public server |
| 21:32.59 | appel11 | he uses it from more locations |
| 21:33.31 | gruvfunk | appel11, if you can't restrict at the firewall, change all your SIP passwords NOW |
| 21:34.11 | appel11 | changed them all restart asterisk rebooted the server |
| 21:34.41 | WIMPy | Don't allowguests. |
| 21:34.43 | appel11 | must say i've already blocked them from calling outside, so the presure is of, and damage is already done |
| 21:35.02 | appel11 | can I see that in the cli? |
| 21:35.06 | appel11 | if that is set? |
| 21:35.11 | gruvfunk | appel11 -> stop asterisk, ensure no processes are running, change all passwords to a complex non-guessable value, then restart asterisk |
| 21:35.24 | koffel | is there a easy way to do asterisk without freepbx? |
| 21:35.32 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 21:35.47 | appel11 | gruvfunk: done that too |
| 21:37.38 | *** join/#asterisk Get_The_Fish (~GTF@c-24-8-50-199.hsd1.co.comcast.net) |
| 21:40.04 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 21:51.17 | *** join/#asterisk phyburn (~phyburn@wsip-70-165-35-234.oc.oc.cox.net) |
| 21:56.08 | TeknoJuce | If my Nortel i2004 IP phone is called black and when I get an inbound call from gv do I want exten => XXXXXXX@gmail.com, n, Dial(SIP/black, 180, D(:1)) |
| 22:02.02 | *** join/#asterisk sahafeez (~sahafeez@4.53.128.211) |
| 22:13.39 | *** join/#asterisk inluck (~inluck@142.162.185.18) |
| 22:14.48 | inluck | I have a script that will transcribe voicemail and makes use of the mailcmd="" directive in the voicemail.conf |
| 22:14.57 | inluck | but I use imap storage for voicemail |
| 22:15.14 | inluck | and am looking for the fuction within asterisk or voicemail module |
| 22:15.25 | inluck | that actually deals with putting the voicemail on the imap server |
| 22:15.56 | inluck | any one have any ideas on where to start? |
| 22:20.14 | bbryant | inluck: well it's certainly in apps/app_voicemail.c |
| 22:20.19 | bbryant | but I'm not sure of the specific function |
| 22:21.51 | inluck | I had found my way to that file |
| 22:21.58 | inluck | and am currently looking through it. |
| 22:22.00 | inluck | Thanks. |
| 22:22.15 | inluck | Might be in over my head on this one. |
| 22:23.25 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 22:28.45 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
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| 22:41.16 | *** join/#asterisk Aut0Exec (~Joe@65.75.65.130) |
| 22:41.32 | Aut0Exec | hi guys.. can someone please help me out wtih some configs for cisco spa3102 |
| 22:43.21 | *** join/#asterisk phyburn (~phyburn@wsip-70-165-35-234.oc.oc.cox.net) |
| 23:02.39 | *** join/#asterisk the_5th_wheel (~edd@webster.cybertek.co.za) |
| 23:04.26 | the_5th_wheel | good day. quick question. How can I escape a sip secret so that it has an @ in it? |
| 23:06.48 | *** join/#asterisk zkn (~zkn@82.131.54.59.cable.starman.ee) |
| 23:09.38 | Aut0Exec | /quit |
| 23:09.39 | Aut0Exec | exit |
| 23:10.26 | zkn | hey, does anyone have an idea how could I get the output by running the following command in linux shell output=$(asterisk -rx 'queue show') ; echo ${output} to look as good simply running asterisk -rx 'queue show' in the shell ? |
| 23:10.52 | zkn | as |
| 23:11.44 | *** join/#asterisk ariel_ (~chatzilla@99-1-236-49.lightspeed.miamfl.sbcglobal.net) |
| 23:13.16 | zkn | i realise this is not a linux shell channel but i want to get output from asterisk to shell |
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