| 00:04.00 | depressed | Is it possible to change the ringing tone asterisk presents incoming callers? |
| 00:04.46 | WIMPy | Yes, but that should hopefully come from your telco. |
| 00:07.01 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 00:12.22 | depressed | That would be the logical solution but even with my Google Voice trunks, my callers are still getting a UK ringing tone when they call in |
| 00:12.54 | depressed | Is it possible Asterisk is playing the ringing tone down the line? |
| 00:13.15 | WIMPy | Are you ansewering the call in your dialplan? |
| 00:14.08 | depressed | Yes |
| 00:14.19 | WIMPy | Don't do that. |
| 00:15.52 | depressed | I tried that but even when the call enters a queue, it still has the UK tone |
| 00:18.37 | WIMPy | Not sure, how to change tones for answered calls. Have you tried to change language? |
| 00:19.04 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 00:19.47 | WIMPy | But perhaps it would be a good idea to play your own recording anyway? |
| 00:30.07 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
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| 01:05.03 | ghostmediapro | no audio on remote sip asterisk cli reports http://pastebin.com/DB1TiDR5 |
| 01:08.40 | p3nguin | Did you configure all the requisite NAT settings? |
| 01:12.04 | *** join/#asterisk d_preston215 (~chatzilla@173-12-4-137-panjde.hfc.comcastbusiness.net) |
| 01:12.11 | d_preston215 | Or experience with DAHDI and redFONE's? |
| 01:12.14 | d_preston215 | Does anybody have experience with DAHDI and dynamic spans? |
| 01:12.32 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 01:12.40 | p3nguin | your questions. |
| 01:12.42 | p3nguin | Just ask |
| 01:14.20 | ghostmediapro | yes all ports are set |
| 01:15.40 | p3nguin | Do you mean, "Yes, all the requisite NAT settings are configured. Additionally, the ports are forwarded," or do you think that only having ports "set" is all that is required? |
| 01:16.19 | ghostmediapro | udp 5060-5070 rtp 10000-20000, iax whatever port it is, don't know off the top of my head |
| 01:16.26 | d_preston215 | Whenever I try to setup dynamic spans in DAHDI 2.5.0, the kernel craps the bed. |
| 01:16.28 | *** join/#asterisk jeffspeff (~jeffspeff@c-76-31-128-113.hsd1.tx.comcast.net) |
| 01:16.32 | ghostmediapro | forwarded |
| 01:16.56 | d_preston215 | For TDMoE, which modules should I have loaded into DAHDI? |
| 01:17.04 | p3nguin | If you've forwarded the ports, now configure asterisk for operating behind NAT with phones behind another NAT. |
| 01:17.32 | ghostmediapro | internal calls are great sip registration and audio good, but remote sip ext register ok, but no audio and it time out after 30 sec |
| 01:17.33 | d_preston215 | I know for 1.6 and DAHDI 2.3.0 its dahdi_dynamic and dahdi_dynamic_ethmf. |
| 01:17.41 | *** join/#asterisk wesphillips (~wphill04@adsl-75-53-131-207.dsl.hstntx.sbcglobal.net) |
| 01:17.47 | *** join/#asterisk Paulius (~Paulius@unaffiliated/paulius) |
| 01:17.56 | p3nguin | I'll repeat: configure the nat settings. |
| 01:18.15 | Paulius | Hey guys. What turnkey Asterisk appliance would you be comfortable with recommending? |
| 01:18.22 | ghostmediapro | sip.conf is configured |
| 01:18.30 | p3nguin | Show me. |
| 01:21.15 | *** join/#asterisk caveat- (~false@gateway/shell/bshellz.net/x-nzekxlawjquavfeg) |
| 01:21.23 | ghostmediapro | http://pastebin.com/e7YcQ4jE |
| 01:21.50 | p3nguin | If that's all you've got, that's not enough. |
| 01:22.25 | ghostmediapro | sip_custom.conf |
| 01:22.29 | p3nguin | You also need nat=yes and directmedia=no if asterisk is behind nat. |
| 01:23.41 | ghostmediapro | where can i find the correct config setting or an example |
| 01:23.53 | p3nguin | I just gave you the settings. |
| 01:24.03 | p3nguin | nat=yes |
| 01:24.12 | p3nguin | directmedia=no |
| 01:24.29 | p3nguin | And if your remote phones are behind other NATs, then you have to have nat=yes and directmedia=no in each peer entry as well. |
| 01:25.54 | jeffspeff | i'm about to pull my hair out trying to get nat working. Here's the scenario -> Phone1: Test1 trying to make call to phone user "jmarcomb". there are two different sites. Phone1 is offsite and behind firewall. asterisk is setup at another site with all the other phones including jmarcomb. I have specified public ip's and opened the ports in the firewall that's onsite with asterisk. While using Phone1 (Test1) I can connect a cal |
| 01:25.56 | jeffspeff | l to jmarcomb and hear audio from jmarcomb; but jmarcomb cannot hear audio from me. Here are the relevant parts of sip.conf http://pastebin.com/CdieeBMR and here is a sip debug http://pastebin.com/0UQfJigW |
| 01:26.19 | *** join/#asterisk wesphillips (~wphill04@adsl-75-53-131-207.dsl.hstntx.sbcglobal.net) |
| 01:26.41 | jeffspeff | i know i'm missing something trivial somewhere; any help is greatly appreciated |
| 01:27.14 | p3nguin | If asterisk is behind nat, you need nat=yes. You have it commented out. |
| 01:27.39 | p3nguin | And if test1 is behind a different nat, it needs to have nat=yes in its entry. |
| 01:27.59 | p3nguin | If it is on the same LAN as asterisk, it does not need nat=yes. |
| 01:28.13 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 01:28.14 | jeffspeff | p3nguin, ty, i was trying it both ways. same results with nat=yes uncommented. Test1 is not on same lan |
| 01:28.46 | p3nguin | What ports did you allow? |
| 01:29.27 | jeffspeff | 5060 through 6000 |
| 01:29.42 | p3nguin | What is running on 5061-6000? |
| 01:30.02 | jeffspeff | afik only asterisk |
| 01:30.02 | p3nguin | And where are the RTP ports? |
| 01:30.03 | p3nguin | SIP is just port 5060. |
| 01:30.20 | ghostmediapro | p3nguin: http://pastebin.com/fS2dZaCh |
| 01:30.24 | jeffspeff | i specified rtp to use 5060 through 6000 |
| 01:30.37 | jeffspeff | does rtp need to start at 5061? |
| 01:31.01 | WIMPy | yes |
| 01:31.27 | p3nguin | ghostmediapro: What's with the public IP address in the localnet setting? |
| 01:32.04 | p3nguin | If SIP is on 5060, you can't run anything else on it on the same bind addresses. |
| 01:32.08 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 01:32.26 | ghostmediapro | sip_custom.conf is what i had to edit right |
| 01:32.26 | jeffspeff | ok, i've changed that |
| 01:32.49 | p3nguin | ghostmediapro: No clue; we don't support FreePBX here. For Asterisk, it's sip.conf. |
| 01:32.50 | jeffspeff | anything else jump out at you as being wrong? |
| 01:36.21 | p3nguin | rtpstart and rtpend are valid in sip.conf now? |
| 01:36.26 | jeffspeff | yes |
| 01:36.35 | jeffspeff | 5061 and 6000 |
| 01:36.39 | p3nguin | checks |
| 01:36.58 | jeffspeff | i'm committing changes and going to test again |
| 01:36.59 | p3nguin | I don't think so. |
| 01:37.20 | jeffspeff | ok, ty waiting for phone to reboot, and i'll test again. |
| 01:37.49 | p3nguin | There's no rtpstart nor rtpend in the sip.conf sample file. But there is a perfectly good rtp.conf just for configuring rtp. |
| 01:38.15 | *** join/#asterisk cyford (~cyford@c-24-30-48-255.hsd1.ga.comcast.net) |
| 01:38.39 | p3nguin | And if test1 is behind a NAT, you need to specify directmedia=no and nat=yes. |
| 01:39.24 | jeffspeff | p3nguin, just opened that file for a peek and found that it specifies rtpstart=10000 |
| 01:39.26 | jeffspeff | <PROTECTED> |
| 01:39.38 | p3nguin | Those are typical default values. |
| 01:39.49 | p3nguin | So open that port range and fix the other one. |
| 01:39.59 | p3nguin | And fix sip.conf to not include those invalid settings. |
| 01:40.22 | p3nguin | SIP is UDP 5060, and RTP is whatever UDP range specified in rtp.conf. |
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| 01:47.58 | *** part/#asterisk wesphillips (~wphill04@adsl-75-53-131-207.dsl.hstntx.sbcglobal.net) |
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| 02:02.42 | jeffspeff | p3nguin, thank you thank you thank you! fixing those rdp ports solved my problem. i have no clue where i'd gotten the 5060-6000 from. |
| 02:03.06 | p3nguin | You probably read it somewhere on the internut. |
| 02:03.06 | jeffspeff | does a jig |
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| 02:53.38 | psykon | What is best practice when you have multiple asterisk boxes behind a firewall that need to talk to different ITSPs? Obviously I can't port forward to *all* of them. Should I utilize some sort of SIP gateway? Maybe OpenSIPS? |
| 02:54.35 | p3nguin | You could use a SIP proxy, or you could just let them talk to the ITSPs directly without a proxy. |
| 02:54.53 | p3nguin | What kind of a firewall do you have? |
| 02:55.42 | psykon | p3nguin, I use a pfSense firewall btw. So should I simply allow a UDP SIP and RTP range and not forward ports? |
| 02:56.21 | p3nguin | pfSense is also a nat server, so that complicates forwarding of ports to multiple systems. |
| 02:56.44 | p3nguin | If you were doing basic firewall on a routable subnet, then it would be easy. |
| 02:56.57 | *** join/#asterisk cstachris (~chrismylo@202.182.147.65) |
| 02:57.18 | p3nguin | It's possible that you don't need to forward any ports at all. |
| 02:58.10 | p3nguin | If pf is capable of keeping state and identifying established and related traffic, it may work just fine without forwarding anything. |
| 02:58.10 | psykon | p3nguin, That is where I get confused. It seems with some providers it is not necessary to *forward* ports, others it is necessary. |
| 02:58.18 | cstachris | hello - does anyone know what "new state Ringing for Notify User 2611 (queued)" means? subscriptions are not working on a telephone |
| 02:58.48 | p3nguin | It doesn't really have anything to do with the providers. |
| 02:58.55 | psykon | p3nguin, What firewall do you recommend? |
| 02:59.12 | p3nguin | pf is a fine product, so pfSense is probably acceptable. |
| 02:59.17 | *** join/#asterisk mintos (~mvaliyav@117.206.19.186) |
| 02:59.35 | psykon | p3nguin, what would *you* personally use? |
| 02:59.39 | p3nguin | I personally use more iptables than pf, but that's just how it worked out for me. |
| 03:00.45 | p3nguin | I only have one internet-facing BSD box running pf. Others are behind Linux-based routers. |
| 03:02.15 | psykon | p3nguin, Wouldn't things be easier with a sip proxy? |
| 03:03.11 | *** join/#asterisk radic (~radic@dslb-178-002-226-204.pools.arcor-ip.net) |
| 03:04.31 | p3nguin | Maybe, if that's what you want to do. |
| 03:06.40 | psykon | p3nguin, I apologize for being so vague. Here is what I have: About 10 Asterisk boxes all using PRIs for PSTN connectivity all on one local subnet. I have a couple of SIP providers and I would like to start to use SIP slowly but I cannot start forwarding to each box. |
| 03:07.15 | p3nguin | And I already said that you may not need to forward anything, so that's where I would start. |
| 03:07.51 | p3nguin | If you need a SIP proxy, use a SIP proxy. |
| 03:08.12 | p3nguin | It's more work than just using Asterisk with the ITSPs directly, so it wouldn't be my first approach. |
| 03:13.08 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 03:19.38 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
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| 03:27.25 | psykon | p3nguin, That leaves me with the question of where to do the carrier registrations. |
| 03:27.54 | p3nguin | I don't understand. |
| 03:28.36 | p3nguin | Are you trying to receive calls from the ITSPs as well? |
| 03:29.06 | psykon | p3nguin, yes |
| 03:29.12 | p3nguin | I think I may have misinterpreted what you meant when you said "talk to." |
| 03:30.25 | p3nguin | You should be able to send calls to the ITSPs without any type of port forwarding, but having multiple asterisk's behind one nat may require your ITSPs to support nat too. |
| 03:31.05 | p3nguin | Think of the case of several phones behind one nat communicating with a single asterisk outside the nat... |
| 03:31.19 | p3nguin | No ports need to be forwarded on the nat where the phones are. |
| 03:31.24 | p3nguin | (usually) |
| 04:15.26 | cstachris | new state Ringing for Notify User 2611 (queued) --- what does (queued) mean? |
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| 06:16.29 | schmidts | morning |
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| 06:28.37 | wdoekes2 | morning |
| 06:31.03 | schmidts | does anyone of you noticed a big amount of hacking attempts from palestina lately? my biggest problem is that they start to change their way of hacking. |
| 06:31.45 | schmidts | i had two cases in the last 3 weeks where they hacked the webinterface of voip phones and get the user data out of there and just register to my servers with the right username + password, so even fail2ban wouldnt help |
| 06:32.21 | schmidts | atleast it was the fault of the customer leaving this phones with a public ip and the default webif pwd but its still something which scares me |
| 06:33.42 | *** join/#asterisk atan (atan@unaffiliated/atan) |
| 06:33.52 | *** join/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it) |
| 06:48.19 | wdoekes2 | haven't noticed more of that. but I use fail2ban on my webinterface as well |
| 06:48.58 | schmidts | wdoekes2 but not on the webinterface of your phones ;) |
| 06:49.26 | wdoekes2 | oh.. didn't see that. heh. no :) |
| 06:50.32 | wdoekes2 | I believe my linksys doesn't disclose the passwords |
| 06:50.53 | schmidts | linksys doesnt show the passwort in plaintext if you take a look at the source code of the webif |
| 06:51.20 | schmidts | Aastra and siemens gigaset does it :( |
| 06:52.36 | irroot | ive had customers web interface nailed recently not only phones if udp 5060 is open looks like port 80 is a target |
| 06:53.07 | irroot | this puts asterisk with a web gui at risk too when 5060/80 is open to net |
| 06:53.19 | schmidts | yes thats true |
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| 07:34.27 | ocx | is there any function in asterisk to sayWord like sayNumber, where the operator will read an english word |
| 07:34.49 | irroot | ocx think its festival text to speach |
| 07:34.54 | irroot | speech |
| 07:37.20 | kaldemar | espeak and flite applications also exist. |
| 07:37.58 | kaldemar | but they are 3rd party modules and the speach is kinda robotic. |
| 07:39.16 | ocx | so you recommend festival? |
| 07:43.43 | kaldemar | just giving you choices, try them out and see what suits you. festival probably sounds the most natural of those, and has more configuration options. |
| 07:46.21 | ocx | thanks |
| 07:53.13 | *** join/#asterisk hetii (~hetii@194.181.154.25) |
| 08:05.45 | ChannelZ | I want a voice that sounds like the old Speak & Spell |
| 08:07.20 | kaldemar | go for flite. :) |
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| 08:17.22 | ChannelZ | I have an old DECtalk app for doing Stephen Hawking |
| 08:26.19 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 08:30.57 | catphish | dectalk would be awesome |
| 08:38.46 | irroot | Stephen Hawking proves upgrades are not needed could have upgraded his voice to sultry sexy [blonde i imagine] but has not ;) |
| 08:39.13 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 08:40.45 | catphish | i think he has upgraded it now |
| 08:40.55 | catphish | quite some time later though |
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| 08:57.02 | ocx | how can i debug a dialplan |
| 08:57.09 | ocx | dialplan -rvvvv |
| 08:57.19 | ocx | any other places to look at some output? |
| 08:58.52 | irroot | check logger.conf include debug there |
| 08:59.08 | irroot | and you can add debug options |
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| 09:01.32 | ocx | coz i am getting status is 'UNKNOWN' in the cli |
| 09:01.37 | ocx | but no more info |
| 09:07.28 | ocx | is there any kind of trace |
| 09:07.30 | ocx | to see |
| 09:07.35 | ocx | of dialplan execution |
| 09:07.51 | cstachris | use NoOp a lot |
| 09:10.13 | ocx | you mean like same => n,NoOP("Step1 executed"); |
| 09:10.14 | ocx | ? |
| 09:12.01 | cstachris | yes |
| 09:12.03 | irroot | when a call happens and you using -vvv then you should see the flow if the context is correct |
| 09:12.20 | cstachris | core set verbose X where X is a number between 0 and 20 |
| 09:12.47 | cstachris | you can output all sorts of variables in a NoOp |
| 09:13.39 | ocx | n,NoOP("Step1 executed"); is not showing anything on the console :/ |
| 09:14.52 | cstachris | exten = 56789,1,NoOp(This NoOp shows the EXTEN being ${EXTEN}) |
| 09:15.09 | ocx | ok without he "" |
| 09:15.15 | cstachris | exten = 56789,n,Hangup |
| 09:15.28 | cstachris | dialplan reload, then dial 56789 and it'll spit out onto your console. |
| 09:15.41 | irroot | ocx dude perhaps no context in channel |
| 09:16.03 | ocx | -- Executing [800@phones:1] Answer("SIP/1001-00000003", "") in new stack == Auto fallthrough, channel 'SIP/1001-00000003' status is 'UNKNOWN' |
| 09:16.07 | ocx | this is what i am seeing |
| 09:16.09 | irroot | cstachris remember dial only works if console driver is loaded |
| 09:16.44 | irroot | ocx "core show dialplan 800@phones" |
| 09:17.39 | ocx | No such command 'core' |
| 09:17.52 | irroot | ocx version |
| 09:18.12 | ocx | asterisk-1.4.41 |
| 09:18.25 | irroot | ah ok |
| 09:18.33 | irroot | ocx "show dialplan 800@phones" |
| 09:19.03 | ocx | localhost*CLI> show dialplan 800@phones [ Included context 'internal' created by 'pbx_config' ] '800' => 1. Answer() [pbx_config] |
| 09:19.17 | cstachris | in 1.4 it's "dialplan show" |
| 09:19.33 | cstachris | oh yeah you've got it there... |
| 09:20.31 | ocx | http://pastebin.com/4he1H3Uu |
| 09:20.33 | ocx | this is my code |
| 09:20.36 | irroot | ocx what you want it todo ?? |
| 09:20.40 | cstachris | sweet |
| 09:20.49 | cstachris | 800,n,NoOp |
| 09:20.50 | ocx | trying to simulate a loop |
| 09:21.01 | cstachris | you've missed the extension number |
| 09:21.17 | ocx | but iwas using same => |
| 09:21.23 | ocx | so no need for the extension number |
| 09:21.27 | cstachris | i've never seen that syntax, excuse me.. |
| 09:21.48 | cstachris | just put exten = 800,n,NoOp so you make progress :) |
| 09:22.08 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 09:23.06 | irroot | i suspect 1.4 does not work with same |
| 09:23.13 | cstachris | probably |
| 09:23.16 | irroot | was it not added in 1.6 ?? |
| 09:23.23 | ocx | it works i actually got it working |
| 09:23.26 | ocx | in other contexts |
| 09:25.15 | *** join/#asterisk Nasga (~Nasga@186.212.10.93.rev.sfr.net) |
| 09:25.22 | cstachris | http://pastebin.com/8ykh0PkM |
| 09:26.17 | cstachris | after you set COUNT=10, in the following line you can put ${COUNT} and it will spit out 10 in the console |
| 09:26.52 | ocx | you only change the same => ? |
| 09:27.09 | cstachris | yep - i don't like that syntax :) |
| 09:27.18 | ocx | let me see if this works |
| 09:28.07 | ocx | -- Executing [800@phones:3] Set("SIP/1001-00000005", "COUNT=10") in new stack == Auto fallthrough, channel 'SIP/1001-00000005' status is 'UNKNOWN' |
| 09:28.20 | cstachris | did you do a "dialplan reload" from the asterisk console |
| 09:28.20 | ocx | so it set thecount but exited afterwards:/ |
| 09:28.24 | ocx | yea sure |
| 09:28.38 | cstachris | ok |
| 09:28.55 | ocx | the noop didnt output too |
| 09:29.30 | ocx | http://pastebin.com/wy12iZ3e |
| 09:30.00 | ocx | sorry the noop output |
| 09:30.02 | ocx | is fine |
| 09:30.08 | ocx | but exited after setting the count |
| 09:31.26 | ocx | exiting on this |
| 09:31.26 | ocx | <PROTECTED> |
| 09:31.46 | cstachris | http://pastebin.com/Rd9fCyFd |
| 09:32.07 | cstachris | reversed the loop, got rid of the :goodbye so it doesn't have the colon. |
| 09:32.39 | cstachris | make sure that works, then you can change it to your own count down rather than count up |
| 09:33.10 | ocx | working fine! |
| 09:33.23 | ocx | i think the problem was in the n(800) |
| 09:33.30 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 09:33.32 | ocx | i included an extra , |
| 09:33.50 | ocx | exten => 800,n(start),GotoIf($[${COUNT} > 0]?:goodbye) |
| 09:33.53 | ocx | should be like that |
| 09:35.20 | cstachris | http://pastebin.com/4he1H3Uu - that is your original - yes, you had an extra n,n, on line 3 |
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| 10:58.48 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 10:59.13 | Dovid | how do I find this in the form of a patch: 2011-08-16 15:06 +0000 [r332021-332026] Matthew Nicholson <mnicholson@digium.com> |
| 11:03.52 | cstachris | Dovid, i would guess to svn co -r 332026 then run svn diff -r 332021 and save the output (just guessing though) |
| 11:10.10 | irroot | Dovid cstachris its slightly more complex than that the range above is accross multiple branches Dovid wants the 1.8 version he need just pull the diff for 332021 |
| 11:12.45 | irroot | Dovid where can i put the diff im pulling it for you |
| 11:13.44 | cstachris | ok |
| 11:14.14 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 11:14.17 | Dovid | irrot: pb |
| 11:14.40 | irroot | could screw it up |
| 11:14.50 | Dovid | iroot: As per the Change log it spans between 332021 and 332026 |
| 11:15.02 | irroot | yeah with gap inbetween |
| 11:15.08 | irroot | its 21 and 26 |
| 11:15.21 | irroot | the missing bits are in other branches |
| 11:15.29 | Dovid | irroot: I pulled that and only saw things relatd to sotresipcause issue |
| 11:15.34 | Dovid | ah ok |
| 11:15.41 | Dovid | so the bestis 1.8.7-rc1 ? |
| 11:16.01 | irroot | that gets my vote for sure |
| 11:16.54 | irroot | ill make patch and commit it to patches folder |
| 11:17.12 | Dovid | iroot: just for my issue ? |
| 11:17.19 | Dovid | so I can then patch 1.8.6.0 ? |
| 11:17.22 | irroot | im feeling nice |
| 11:17.30 | Dovid | iroot: thanks a lot |
| 11:18.16 | Dovid | it seems the change is only in: 332026 |
| 11:18.41 | Dovid | see: http://pastebin.com/aJFs8mXi |
| 11:18.50 | irroot | 331955 is previous commit you need to do 331955:332026 |
| 11:20.13 | Dovid | iroot: Why do i need from 331955 ? |
| 11:20.45 | irroot | Dovid 0 is a number too |
| 11:21.01 | Dovid | i guess this patch since on other settings that were made between 1.8.60 and 332026 ? |
| 11:21.14 | irroot | http://svnview.digium.com/svn/asterisk/branches/1.8/?view=log&pathrev=332026 |
| 11:21.33 | Dovid | looking |
| 11:22.47 | Dovid | meh. i am so foncused |
| 11:23.20 | Dovid | gona get the rc-1 |
| 11:23.44 | irroot | http://svnview.digium.com/svn/asterisk/team/irroot/patches/storesipcause-1.8.patch |
| 11:24.22 | irroot | Dovid ^^ |
| 11:24.49 | Dovid | iroot: I can apply that to a 1.8.6.0 system ? |
| 11:24.56 | irroot | yip |
| 11:25.21 | irroot | should be fine |
| 11:25.23 | Dovid | iroot: Doing that now. |
| 11:25.32 | Dovid | **should** be ? ;) |
| 11:25.40 | Dovid | or u think rc-1 is a better idea? |
| 11:26.13 | Dovid | i am just nervos about other issues. i want to minimize the problems |
| 11:26.46 | irroot | yeah i understand |
| 11:27.06 | irroot | have 70+ sites rolled out with 7-rc1 |
| 11:27.11 | irroot | im using 10 at the office |
| 11:27.42 | Dovid | iroot: I still need to catch up on 1.8 b4 i am @ 10. |
| 11:27.48 | Dovid | i will try rc-1 and pray |
| 11:28.10 | irroot | Dovid do shout im happy to assist |
| 11:28.27 | Dovid | iroot: thanks |
| 11:28.54 | irroot | oej lunch time |
| 11:32.02 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 11:36.07 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip24-56-6-80.ph.ph.cox.net) |
| 11:42.02 | irroot | Dovid 7-rc2 is in svn but not seen a release so check in little bit |
| 11:43.34 | Dovid | iroot: What does it have ? |
| 11:45.34 | irroot | Dovid fix for ilbc , core level suport for meetme, problem with libpri detection for old libpri, cut and paste regresion |
| 11:47.09 | Dovid | core level suport for meetme == ? |
| 11:47.38 | irroot | its prolly related to the build system |
| 11:48.18 | Dovid | ok |
| 11:49.06 | irroot | remember -rc2 is problems from -rc1 unless agreed to and notified as with ilbc |
| 11:51.58 | Dovid | iroot: what issues were crated by rc1 ? |
| 11:52.04 | Dovid | and where can I see the changelog? |
| 11:53.15 | irroot | http://svnview.digium.com/svn/asterisk?view=revision&revision=336635 |
| 11:57.32 | *** join/#asterisk Fritz09 (~Fritz09@46.115.1.149) |
| 11:57.45 | *** join/#asterisk pierrep (~pierre@85-171-104-225.rev.numericable.fr) |
| 11:57.47 | pierrep | hi |
| 11:58.29 | pierrep | our sip phone system works fine for 2 minutes after each script restart |
| 11:58.58 | pierrep | after that we got an unknown number when we call our number |
| 11:59.08 | pierrep | any idea where that could come from |
| 11:59.11 | pierrep | ? |
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| 12:10.55 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 12:12.29 | atan | pierrep, nat? |
| 12:14.24 | ocx | maybe you can try using iax |
| 12:14.32 | ocx | instead of sip |
| 12:15.05 | *** part/#asterisk cstachris (~chrismylo@202.182.147.65) |
| 12:17.45 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 12:18.25 | leifmadsen | pierrep: can't tell without looking at debug information such as a trace and console log |
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| 12:55.23 | *** join/#asterisk Rienzilla (rien@sinas.rename-it.nl) |
| 12:55.59 | Rienzilla | hey everyone. I have a question. When I get an incoming call from nmy sip provider it tells me " chan_sip.c:20276 handle_request_invite: Call from '' to extension '31413230005' rejected because extension not found in context 'default'.", but I specified another context in sip.conf |
| 12:56.06 | Rienzilla | what may I be doing wrong? |
| 12:58.50 | *** join/#asterisk turtlefence (~turtlefen@c58-111-144-182.thorn2.nsw.optusnet.com.au) |
| 12:59.11 | *** part/#asterisk turtlefence (~turtlefen@c58-111-144-182.thorn2.nsw.optusnet.com.au) |
| 12:59.14 | leifmadsen | Rienzilla: sounds like it's not matching the correct peer in your sip.conf file |
| 13:00.16 | Rienzilla | there's not many of them |
| 13:00.23 | leifmadsen | ok... |
| 13:00.27 | leifmadsen | more information required |
| 13:00.43 | leifmadsen | I'd start with some sort of debug information and the configuration you're using (no passwords) |
| 13:00.45 | leifmadsen | use a pastebin |
| 13:00.55 | Rienzilla | yes I will, sec |
| 13:01.50 | Rienzilla | hmm |
| 13:02.05 | *** join/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it) |
| 13:02.53 | Rienzilla | hmm wait, will register = <blah> under [general] instead of the correct peer cause this behaviour? |
| 13:03.07 | leifmadsen | register is not for authentication of incoming calls |
| 13:03.14 | leifmadsen | all it does is tell the other end where you exist on the network |
| 13:03.51 | Rienzilla | ok |
| 13:03.54 | Rienzilla | hold on |
| 13:04.45 | irroot | for dynamic ips you can only receive calls once registered for fixed ips this is no problem |
| 13:05.23 | *** join/#asterisk serafie (~erin@nat/digium/x-indmnnkbbgerzuud) |
| 13:07.31 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-pgokadmracvlfsbc) |
| 13:07.33 | irroot | serafie o/ |
| 13:07.42 | irroot | +1 to mjordan lo peep |
| 13:08.53 | Rienzilla | http://pastebin.com/524bwrNX |
| 13:08.58 | Rienzilla | leifmadsen: ^^ |
| 13:09.15 | mjordan | irroot: what'd I do? |
| 13:09.38 | irroot | you joined :P top 'o the morning .... |
| 13:11.34 | *** join/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it) |
| 13:18.19 | mjordan | morning :-) |
| 13:23.48 | Rienzilla | leifmadsen: I can't figure it out. It seems that it sometimes works, and sometimes doesn't. With the exact same configuration |
| 13:24.50 | leifmadsen | Rienzilla: sounds like they are using multiple systems/IPs then to send the call, and it is matching on only the one IP |
| 13:25.23 | Katty | nff ergg zzzffffllll *zombies in* |
| 13:25.28 | leifmadsen | Rienzilla: your register line goes in the [general] section not the peer definition |
| 13:25.34 | tzanger | hands zombie katty a coffee |
| 13:25.41 | Rienzilla | that's possible. Their proxy IP points to multiple IP's |
| 13:25.44 | Katty | noms. |
| 13:26.03 | Rienzilla | so ehm, how would I solve that on my side? |
| 13:26.10 | Rienzilla | ehm |
| 13:26.14 | Rienzilla | proxy ip= proxy hostname |
| 13:27.01 | leifmadsen | Rienzilla: probably need to add peers for all your hosts, or match on the username correctly |
| 13:27.11 | leifmadsen | (DID number or whatever they use |
| 13:28.01 | *** part/#asterisk hetii (~hetii@194.181.154.25) |
| 13:31.59 | ocx | does anyone know of any tiny hardware on which i can install 2 pci cards (1 for fxo/fxs and ethernet for voip) i would like to build that system as an asterisk system |
| 13:32.03 | ocx | but i want it very small |
| 13:32.14 | ocx | not like the normal desktop motherboard and power supply |
| 13:33.06 | Rienzilla | leifmadsen: hmmmm... |
| 13:33.45 | leifmadsen | ocx: just look for pretty much any microatx system or whatever that has your requirements |
| 13:34.10 | irroot | ocx we use a USB only atom box but no pci slots can use SPA3102/.... |
| 13:34.12 | ocx | hmmm but would that be compatible with linux kernel? |
| 13:34.42 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:36.44 | irroot | ocx yes indeed |
| 13:36.49 | Rienzilla | leifmadsen: so if I have a peer entry for every host I can expect a call from, I'm fine? |
| 13:36.58 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:37.05 | leifmadsen | Rienzilla: if it works, sure |
| 13:39.27 | Rienzilla | well, I'm not sure if it will. Uptil now it was randomly either working or not :) |
| 13:41.01 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:44.59 | Katty | hi _Corey_ |
| 13:45.12 | _Corey_ | good morning :) |
| 13:45.58 | Katty | tzanger: you going to join the xmas card exchange this year? |
| 13:48.22 | Rienzilla | /.3~3~4~4~3~/o=k[ |
| 13:48.49 | irroot | Katty you doing a asterisk lingerie calendar ?? |
| 13:51.33 | ocx | any fxo/fxs PCI Express Mini Card available or are they all standard size? |
| 13:52.30 | Katty | irroot: probably not |
| 13:53.21 | p3nguin | Did you even consider it? |
| 13:54.58 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:56.03 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 13:56.16 | anonymouz666 | I am in |
| 13:56.22 | brad_mssw | are there any plans to include dahdi-linux updates in the binary ubuntu packages? https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages .... ubuntu 10.04LTS includes old 2.2.1 |
| 13:56.24 | p3nguin | That's what she said. |
| 13:56.54 | ocx | any good reviews on the VA400p hardware? |
| 13:57.17 | ocx | Va400p |
| 13:57.26 | Katty | anonymouz666: you joining the xmas card exchange this year? |
| 13:58.42 | anonymouz666 | no, tell me more |
| 13:58.56 | Katty | err well, there's not a whole lot to tell |
| 13:59.02 | Katty | it's just a list of peeps and their addresses |
| 13:59.09 | Katty | and you send cards to the ones you know and talk to lots |
| 13:59.48 | p3nguin | And if people like you, you might even get a card yourself. |
| 13:59.50 | anonymouz666 | Katty: do you wanna get a xmas card from Rio - Brazil? |
| 14:00.04 | Katty | already got south africa, why not! |
| 14:00.19 | Katty | if you want in, lemme know |
| 14:00.36 | Katty | hugs irroot |
| 14:01.04 | irroot | Katty hehehe {{{}}} ps i offered to do a lingerie shoot but no one took up the offer :P |
| 14:01.22 | Katty | maybe it's cause of where you live |
| 14:01.32 | Katty | it's kind of far away |
| 14:01.36 | p3nguin | haha |
| 14:01.44 | malcolmd | ocx: buy a tdm410 from digium instead. it supports asterisk. |
| 14:01.45 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 14:01.46 | anonymouz666 | irroot: I asked you more than once (sorry I forgot), but how to get your t.38 gw stuff in 1.8? |
| 14:02.03 | p3nguin | That's what I keep telling myself about no one wanting to do a lingerie shoot with me. |
| 14:02.09 | p3nguin | It's 'cause of where I live. |
| 14:02.11 | irroot | anonymouz666 apply the patch or pull the t38 branch what you prefer ? |
| 14:02.15 | ocx | malcolmd: is it a mini pci? |
| 14:02.18 | ocx | i need a mini pci |
| 14:02.25 | ocx | need to build a micro atcx |
| 14:02.26 | ocx | atx |
| 14:02.29 | p3nguin | mini pci, or mini pci-e? |
| 14:02.30 | ocx | so the smaller the better |
| 14:02.33 | malcolmd | nope, but neither is that va400p |
| 14:02.44 | ocx | saw this now |
| 14:02.45 | ocx | http://www.openvox.cn/en/products/fxofxs-cards/a400m.html?page=shop.product_details&flypage=flypage.tpl&product_id=20&category_id=1 |
| 14:02.46 | anonymouz666 | irroot: I am using the 1.8.7.0-rc1 today, with patches that still are not merged in |
| 14:02.56 | anonymouz666 | in 1.8.8.0 all of them will be |
| 14:03.16 | anonymouz666 | so I think through patche will be better |
| 14:04.18 | irroot | then pull the branch "svn co http://svn.digium.com/svn/asterisk/team/irroot/t38gateway-1.8" |
| 14:04.33 | Katty | file: ping |
| 14:04.51 | serafie | irroot: mornin'! |
| 14:05.05 | anonymouz666 | irroot: this will return just the diff? |
| 14:05.14 | anonymouz666 | or the whole asterisk and then a diff -ru myself ? |
| 14:05.30 | Katty | raden: ping |
| 14:05.36 | irroot | its based on branches/1.8 "1.8.8" its the whole thing |
| 14:05.36 | Katty | infobot: seen eppigy |
| 14:05.41 | infobot | eppigy <~eppigy@c-76-105-72-69.hsd1.ga.comcast.net> was last seen on IRC in channel #asterisk, 413d 22h 20m 43s ago, saying: 'pretty much what i expected though i guess'. |
| 14:05.58 | *** join/#asterisk master_of_master (~master_of@p57B5410F.dip.t-dialin.net) |
| 14:06.02 | irroot | infobot: seen irroot |
| 14:06.03 | infobot | irroot is currently on #asterisk-dev (8h 42m) #asterisk (8h 42m) #asterisk-bugs (8h 42m). Has said a total of 78 messages. Is idling for 1s, last said: 'infobot: seen irroot'. |
| 14:06.19 | irroot | ~beer |
| 14:06.19 | infobot | ACTION has disconnected (Read error: 99 (Connection reset by beer)) |
| 14:06.31 | anonymouz666 | irroot: it is missing something from the 10 version? |
| 14:06.31 | Katty | jeebus, over a year |
| 14:07.04 | _Corey_ | irroot: You killed the bot! |
| 14:07.10 | leifmadsen | no he didn't :)( |
| 14:07.14 | leifmadsen | infobot: ohai |
| 14:07.16 | _Corey_ | lol |
| 14:07.17 | Katty | infobot: seen katty |
| 14:07.19 | infobot | katty <~Katty@mail.copi-rite.com> was last seen on IRC in channel #asterisk, 2s ago, saying: 'infobot: seen katty'. |
| 14:07.43 | _Corey_ | I did see the 'beer'... |
| 14:07.48 | irroot | anonymouz666 the fax bits are in sync with 10 |
| 14:07.57 | kaldemar | brad_mssw: you'd have to ask ubuntu about the packages they include. |
| 14:08.13 | anonymouz666 | irroot: amazing |
| 14:08.27 | anonymouz666 | so it's time to make a diff myself and apply to my version |
| 14:08.31 | anonymouz666 | this is needed |
| 14:09.09 | irroot | http://svnview.digium.com/svn/asterisk/team/irroot/patches/t38gateway-1.8.patch |
| 14:09.29 | irroot | anonymouz666 the above is a patch of the tree |
| 14:10.08 | anonymouz666 | ooh, nice! |
| 14:16.33 | Katty | kaldemar: you joinin the xmas card exchange this year? |
| 14:18.35 | *** join/#asterisk grEvenX (~even@2.151.6.51.tmi.telenormobil.no) |
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| 14:22.58 | kaldemar | Katty: haven't even heard of it. :) what is it? |
| 14:25.08 | Katty | why has no one heard of the xmas card exchange? we've been doin it for years lol |
| 14:25.23 | Katty | kaldemar: it's just a list of peeps, and you send cards to the ones you know and talk to |
| 14:25.36 | Katty | kaldemar: and likely will get some back. i send cards to everyone on the list |
| 14:25.57 | d_preston215 | Does DAHDI 2.5 even have support for Dynamic spans?\ |
| 14:27.50 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:28.13 | d_preston215 | Does DAHDI 2.5 have any issues with ACPI? |
| 14:28.43 | WIMPy | Dynamic spans? |
| 14:29.00 | d_preston215 | Yes. |
| 14:29.17 | WIMPy | What's that? |
| 14:30.22 | kaldemar | Katty: oh ok. i'm a bit lazy in that sense. :| |
| 14:30.30 | Katty | okie dokie |
| 14:30.42 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:30.44 | Katty | kaldemar: i may ask you again, my memory is awful ;) |
| 14:30.56 | d_preston215 | I doubt you would know what a redFONE is then. |
| 14:32.35 | WIMPy | I don't know much about them, no. |
| 14:33.38 | kaldemar | WIMPy: TDMoE or loopbacks. |
| 14:33.54 | d_preston215 | Yeah. |
| 14:34.36 | *** join/#asterisk VoipForces (~Adium@69.165.199.195) |
| 14:34.36 | WIMPy | So it's just about adding/removing channels at runtime? |
| 14:35.20 | VoipForces | Hi all, anyone ever had an issue with a Cisco SPA303 where the hone works, but when you hangup from a call all LED turn orange and you need to reboot the phone to get it working again ? |
| 14:37.25 | kaldemar | WIMPy: no, running TDM over ethernet or creating loopback spans for example. |
| 14:37.40 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:37.55 | WIMPy | Why is that called dynamic? |
| 14:41.25 | tzanger | Katty: I probably should, shouldn't I |
| 14:41.38 | Katty | if you want to, you should (= |
| 14:43.43 | kaldemar | WIMPy: probably because the span doesn't run on any specific telephony hardware. who knows. |
| 14:46.09 | *** join/#asterisk psilikon (~joel@rrcs-24-73-84-158.se.biz.rr.com) |
| 14:48.31 | raden | Katty, Pong |
| 14:49.13 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 14:50.55 | WIMPy | It's somewhat misleading. |
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| 14:53.12 | *** join/#asterisk corretico (~luis@200.12.40.18) |
| 14:53.25 | Katty | raden: you joining the xmas card exchange this year? |
| 14:54.20 | *** join/#asterisk jkroon (~jkroon@dsl-242-10-94.telkomadsl.co.za) |
| 14:54.26 | jkroon | leifmadsen, are you online? |
| 14:54.40 | p3nguin | ~seen leifmadsen |
| 14:54.51 | infobot | leifmadsen <~Leif@asterisk/documenteur-extraordinaire/blitzrage> was last seen on IRC in channel #asterisk, 47m 37s ago, saying: 'infobot: ohai'. |
| 14:54.52 | jkroon | was hoping to get some help with the fdopen issue ... it's affecting four of my clients now and I just can't figure it out. |
| 14:55.12 | leifmadsen | what is the 'fdopen' issue? |
| 14:56.48 | p3nguin | Interesting how upgrading asterisk apparently breaks mutt. |
| 14:56.51 | p3nguin | Couldn't lock /var/lib/asterisk/sent |
| 14:57.06 | p3nguin | It doesn't make any sense to me. |
| 14:57.54 | irroot | jkroon yeah i looked at it and have no clue |
| 14:58.47 | p3nguin | I don't see how there's even an association between the two softwares. |
| 14:59.15 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 15:01.07 | *** join/#asterisk corretico (~luis@200.12.40.18) |
| 15:01.19 | irroot | leifmadsen blocker for 1.8.8 ?? ASTERISK-18453 jkroon |
| 15:02.40 | *** join/#asterisk corretico (~luis@200.12.40.18) |
| 15:06.51 | irroot | jkroon is it possible you running out of file descriptors |
| 15:09.22 | Qwell | HTTP FDs stick around for a while, I do believe. |
| 15:10.35 | irroot | also if the sessions are not closed i guess it could add to the time |
| 15:23.09 | *** join/#asterisk Polis_ttt (~lasse@irc.mussla.se) |
| 15:24.40 | Katty | hi Qwell |
| 15:24.59 | Qwell | ohai |
| 15:25.05 | Katty | everytime i move my shoulder, it pops |
| 15:25.07 | Katty | is that normal? |
| 15:25.12 | Qwell | Don't do that then. |
| 15:25.25 | Qwell | it's not abnormal. I pop a lot. |
| 15:25.31 | Katty | k |
| 15:25.34 | irroot | Katty you a zombie ?? seen that happen with them so normal |
| 15:26.06 | Katty | no i've had my morning caffeine, so not a zombie anymore |
| 15:26.07 | leifmadsen | irroot: no, not a blocker imho |
| 15:28.04 | irroot | indeed unless we can eliminate the obvious |
| 15:30.04 | *** join/#asterisk d_preston215 (~chatzilla@173-12-4-137-panjde.hfc.comcastbusiness.net) |
| 15:31.07 | navaismo | hi, yesterday someone mention something about asterisk dial IPs when some peer its not reachable, how can i disabled that function? |
| 15:31.53 | p3nguin | Don't dial peers that do not exist. |
| 15:32.26 | Katty | that sounds like relationship advice |
| 15:33.02 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:34.30 | navaismo | mm the strange thing this happens in a queue |
| 15:34.34 | navaismo | all members exist |
| 15:35.34 | p3nguin | I'd rather dial local channels as opposed to dialing devices directly from the queue. |
| 15:37.57 | kaldemar | navaismo: that was not about a peer being unreachable. the person was dialing something that was not defined as a peer in sip.conf, and hence it was interpreted as an ip address. |
| 15:38.28 | *** join/#asterisk Fritz09 (~Adium@pop1-2551.catv.wtnet.de) |
| 15:38.43 | navaismo | ok im confused now, the queue number its 10000 then it dial to 1000-1009 |
| 15:38.50 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 15:39.01 | navaismo | and i see a lot of sip_xmit messages and then the call drops |
| 15:39.32 | p3nguin | A queue member of 10000 isn't valid. |
| 15:39.44 | p3nguin | Wait, disregard. |
| 15:39.55 | p3nguin | I read your statement wrong. |
| 15:40.12 | p3nguin | What do you mean it dials 1000-1009? |
| 15:40.47 | p3nguin | The member is Local/1000? |
| 15:40.59 | leifmadsen | navaismo: it's not a function that you can disable -- just check if the peer exists first |
| 15:43.06 | navaismo | p3nguin the queue dial peers 1000 to 1009 |
| 15:43.21 | p3nguin | So you mean the members are SIP/1000 etc? |
| 15:43.26 | navaismo | yes |
| 15:43.29 | p3nguin | (1035.33) <p3nguin> I'd rather dial local channels as opposed to dialing devices directly from the queue. |
| 15:43.32 | p3nguin | Consider this. |
| 15:44.10 | navaismo | leifmadsen ok ill check it that |
| 15:44.28 | p3nguin | When you use local channels instead of devices directly, you can do useful things such as checking if the peer exists before trying to dial it. |
| 15:44.41 | navaismo | interesting |
| 15:45.06 | leifmadsen | OMG REALLY?! |
| 15:45.18 | navaismo | so |
| 15:45.43 | navaismo | i need to set the menber like local/1000@context? |
| 15:45.53 | p3nguin | That's what I was told, anyway; I personally don't know anything. |
| 15:45.55 | p3nguin | Correct. |
| 15:46.25 | irroot | p3nguin navaismo will need a custom device state for this purpose im thinking |
| 15:46.32 | p3nguin | Why? |
| 15:46.44 | leifmadsen | irroot: you mean interface state |
| 15:47.08 | irroot | yes for queues to track it |
| 15:47.32 | irroot | interface , interface_state the local points to a legit interface state |
| 15:47.45 | p3nguin | I have a context called queue-devices, which has an extension pattern that pulls the device names from astdb, checks if they exist, and then dials the phone with a special ring option. It works fine. |
| 15:48.21 | irroot | cool must look into this |
| 15:48.55 | p3nguin | Then the queue members are like Local/762@queue-devices |
| 15:49.38 | p3nguin | If 762 matches to a peer in the astdb, it'll be checked for existence and called if it exists. |
| 15:50.32 | p3nguin | This concept is based on the same concept used for calling phones normally. |
| 15:51.13 | p3nguin | (just a special context for queues to set custom options that are not used when calling normally) |
| 15:52.02 | p3nguin | If you have off-hook queue agents, this isn't necessary -- you'd use chan_agent instead. |
| 15:52.40 | p3nguin | For a small-to-medium office, I prefer on-hook agents. |
| 15:52.42 | Qwell | chan_agent no longer exists* |
| 15:52.45 | Qwell | (*in my mind) |
| 15:52.56 | p3nguin | What has replaced it? |
| 15:54.45 | p3nguin | I only ever used chan_agent in a couple situations, and it seemed to work okay. I imagine replacing it with other available tools would make things much more complicated. |
| 15:58.37 | navaismo | :S |
| 15:59.01 | navaismo | im very confused now |
| 15:59.52 | brad_mssw | are there any plans to include dahdi-linux updates in the binary ubuntu packages? https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages .... ubuntu 10.04LTS includes old 2.2.1, and the asterisk/digium ubuntu repo doesn't include a dahdi-linux update, but _does_ provide other things like asterisk 1.8 |
| 16:00.17 | Qwell | brad_mssw: Those packages use DAHDI from upstream. |
| 16:00.52 | Qwell | pabelanger: Making our own versions of that stuff really aught to be high on the list though. |
| 16:03.37 | brad_mssw | Qwell: yeah, it'd be awfully convenient if the asterisk-provided repo for ubuntu included dahdi-linux, so us ubuntu people didn't have to rely on ubuntu's versions which are a couple years old |
| 16:04.14 | brad_mssw | Qwell: or obviously, compile it ourselves and thus can't rely on DKMS to auto-rebuild for kernel updates, etc |
| 16:04.32 | pabelanger | Right, ATM we are only package Asterisk and depend on any OS package, so in your case DAHDI |
| 16:04.50 | pabelanger | s/age/aging/ |
| 16:08.31 | p3nguin | navaismo: What are you having trouble with? I'll do my best to clarify for you. |
| 16:10.18 | Katty | dear caffeine, i love you, but why do you always make me have to pee constantly!!! argh... |
| 16:12.30 | brad_mssw | I assume running dahdi 2.2.1 would be considered a 'bad' thing these days, right? I've been having a PRI issue where I need to restart every weeks or so to bring the pri back online and get "PTP MDL can't handle error of type A" and "MDL-ERROR (A): Got supervisory frame with p_f bit set to 1 in state 7" errors in the asterisk console reported when things go south |
| 16:14.29 | pabelanger | I don't think it is a 'bad' thing to run 2.2.1. Actually, I wouldn't even know what the changes are between 2.2.1 and 2.5 |
| 16:19.48 | anonymouz666 | I am getting Digium spam :) "Can you spare 3 minutes to share your thoughts on PBX systems?" more than 3 times today :P |
| 16:20.23 | p3nguin | Don't create four or more accounts with Digium. |
| 16:20.28 | drift- | p3nguin! |
| 16:21.16 | anonymouz666 | p3nguin: I don't even remember that I create one! :D |
| 16:22.46 | *** join/#asterisk hmmhesays (~hmmhesay@174-126-194-60.cpe.cableone.net) |
| 16:26.42 | *** join/#asterisk navaismo (~navaismo@187.170.13.99) |
| 16:26.58 | navaismo | p3nguin sorry LAN issues |
| 16:27.13 | p3nguin | anonymouz666: I have one account, and I got one email. |
| 16:29.16 | *** join/#asterisk dms (~dms@nat/digium/x-qectwhokclhrcdhx) |
| 16:30.21 | irroot | anonymouz666 yeah digum had me till there was a ipad on offer was going to send to customers and staff but its safely in my trash folder |
| 16:34.54 | n3hxs | Katty, you need to take your caffeine in dry solid form. ;) |
| 16:39.07 | navaismo | p3nguin this a part of the output http://pastebin.com/4MNSZWYn in there tha call isnt reach the queue yet its the welcome message |
| 16:41.19 | Katty | n3hxs: pff |
| 16:41.34 | p3nguin | navaismo: I'd like to see s@from-pstnA from extensions.conf. |
| 16:42.12 | d_preston215 | Starting to think that DAHDI is just a crock of crap for dynamic spans. |
| 16:42.31 | d_preston215 | Or just have dropped support for redFONE's in recent versions. |
| 16:43.00 | navaismo | ok wait me a minute |
| 16:46.30 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 16:49.06 | d_preston215 | Is it an outright IRQ issue when I get this: |
| 16:49.08 | d_preston215 | line 0: Unable to open master device '/dev/dahdi/ctl' |
| 16:49.30 | navaismo | p3nguin: here is the context: http://pastebin.com/AvuaiZ2w |
| 16:50.11 | *** join/#asterisk bchia (~Adium@nat/digium/x-bujrbeskthnjfouj) |
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| 16:51.52 | p3nguin | navaismo: There's nothing SIP in your dial plan, so why is chan_sip in your log output? |
| 16:52.39 | navaismo | I guess its a delayed call |
| 16:52.43 | navaismo | from the queue |
| 16:53.53 | navaismo | the queue looks: http://pastebin.com/szdCA0SD dont pay attention to the gotoif |
| 16:54.43 | blizzow | I think I have a phone that keeps registering at the same extension as one that's already registered. How do I find the IP of the phone that's been registering at that extension? |
| 16:55.14 | p3nguin | navaismo: How does the call get to that context/extension? The goto in the other context takes it there? |
| 16:55.46 | p3nguin | blizzow: Phones register as devices. What kind of device is the phone? SIP, perhaps? |
| 16:56.04 | navaismo | yep exten => 1,1,Goto(menuMaltese,s,1) |
| 16:56.05 | navaismo | exten => 2,1,Goto(menuEnglish,s,1) |
| 16:56.09 | p3nguin | okay |
| 16:57.43 | blizzow | p3nguin: I'm not sure, it may be a softphone or a hardphone. |
| 16:58.42 | p3nguin | blizzow: If it is a sip phone, sip show peer <phone's name> |
| 17:05.55 | *** part/#asterisk ghostmediapro (~Administr@71.23.182.164) |
| 17:12.11 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:12.52 | d_preston215 | What IRQ issues can you run into with DAHDI? |
| 17:14.07 | WIMPy | Shared IRQs. What hardware? |
| 17:19.11 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 17:21.10 | *** join/#asterisk MarKsaitis (~MarKsaiti@85-189-201-83.tmbsystems.managedbroadband.co.uk) |
| 17:25.34 | jkroon | irroot, no. |
| 17:26.05 | jkroon | leifmadsen, fdopen issue: https://issues.asterisk.org/jira/browse/ASTERISK-18453 |
| 17:26.36 | jkroon | available now for some testing if you're keen. |
| 17:29.58 | Rienzilla | http://nieuw.begrotingswijzer.nl/ |
| 17:30.18 | Rienzilla | argh |
| 17:30.22 | Rienzilla | disregard that |
| 17:32.09 | *** join/#asterisk tonsofpcs (~tonsofpcs@cpe-69-205-240-64.stny.res.rr.com) |
| 17:33.16 | *** join/#asterisk irroot (~irroot@41.49.174.238) |
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| 17:58.52 | tzanger | Rienzilla: that looks rot13'd |
| 17:58.53 | tzanger | heh |
| 17:59.00 | jkroon | The previous reload command didn't finish yet |
| 17:59.25 | jkroon | why would a sip reload get stuck? it seems to be getting involved in a DNS resolution situation from the looks of it ??? |
| 18:01.42 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 18:07.53 | ChannelZ | could be |
| 18:09.17 | jkroon | ChannelZ, ok ... |
| 18:09.28 | jkroon | that doesn't help :( |
| 18:09.35 | jkroon | gdb here we come ... |
| 18:12.47 | *** join/#asterisk irroot (~irroot@41.52.147.110) |
| 18:13.18 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 18:18.00 | jkroon | http://pastebin.com/sYqnijBm ... that is the all threads stack trace for sip reload lockup ... |
| 18:18.10 | jkroon | ast 1.8.6.0 - any chance someone could take a peek please? |
| 18:18.34 | irroot | jkroon ill peek ps try set the fdlimit higher for the fdopen bug |
| 18:18.45 | jkroon | already at 4096 |
| 18:18.50 | jkroon | on an idle system :p |
| 18:19.42 | irroot | you got a core show locks ?? or debug off |
| 18:19.44 | jkroon | got about 30 files open at any given point in time. |
| 18:19.53 | jkroon | debug off unfortunately. |
| 18:20.02 | jkroon | can't reproduce the fdopen bug with it on. |
| 18:21.10 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 18:21.58 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 18:22.18 | irroot | jkroon see thread 38 seems to be the bussiness end |
| 18:22.31 | irroot | looks like its reading sip.conf |
| 18:22.51 | jkroon | _IO_file_underflow ... ? |
| 18:22.57 | jkroon | bug in the config loader? |
| 18:23.10 | irroot | yeah thats internal in libc |
| 18:23.34 | jkroon | are you pointing at a potential libc bug ?!? |
| 18:23.54 | irroot | this should not happen you need to see why this happen |
| 18:24.08 | irroot | could also be related to fdproblem |
| 18:24.24 | jkroon | agreed. |
| 18:25.52 | jkroon | SIGABRT! |
| 18:25.58 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 18:26.15 | jkroon | 0x00007ffff74768a5 in raise () from /lib64/libc.so.6 |
| 18:28.25 | jkroon | think i found the problem! |
| 18:28.38 | jkroon | sip peers too soon after a reload, so the config is still busy loading the peers. |
| 18:28.48 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 18:28.59 | jkroon | in _sip_show_peers the count is taken before taking a lock, and peerarray then overflows! |
| 18:29.25 | jkroon | this could potentially explain both the peers going MIA as well as the fdopen bug ... |
| 18:30.15 | jkroon | no, i don't think so. irroot chan_sip.c |
| 18:30.25 | *** join/#asterisk Felix1125 (~john@nat/digium/x-swmtsxwprpubpxpg) |
| 18:30.49 | jkroon | objcount is taken on the peers ao2 container (no lock) at line 16109 |
| 18:31.07 | jkroon | ast_calloc then allocated peerarray at line 16115 |
| 18:31.15 | jkroon | still no lock. |
| 18:32.07 | jkroon | iterator initiates around line 16141, what would happen if the number of peers _increase_ during this period from another thread? |
| 18:32.31 | Qwell | looks at Felix1125 |
| 18:32.38 | jkroon | if i'm right total_peers will end up being more than the allocated number of pointers, resulting in overflowing the buffer. |
| 18:33.08 | *** join/#asterisk frem (41b769ca@gateway/web/freenode/ip.65.183.105.202) |
| 18:33.52 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 18:34.21 | frem | So, I've got everything set up with AsteriskNOW, and it almost works perfectly... but if you're connecting to Asterisk and you're not on the internal network, external calls are intermittently choppy. |
| 18:34.39 | Felix1125 | ...puzzled by Qwell's action. |
| 18:34.40 | leifmadsen | sounds like a network issue |
| 18:34.40 | Qwell | frem: That would be an issue with your/their connection. |
| 18:34.54 | Qwell | Felix1125: Who are you? It's not often I see a name coming from our NAT that I don't recognize. |
| 18:35.04 | irroot | jkroon remember the ao2 functions lock the containers and have versioning |
| 18:35.18 | Felix1125 | I'm an Intern at Digium. |
| 18:35.31 | tzanger | heh |
| 18:35.33 | Qwell | Felix1125: ahh, okay - I was thinking that might be the case, but figured it was too early still. |
| 18:35.33 | tzanger | "behind a successful man is his woman. Beind the fall of a successful man is usually another woman." |
| 18:35.43 | jkroon | easy enough to add a quick check for the theory. |
| 18:35.48 | frem | Qwell: We have 3Mbps fiber, they have high speed business cable. :-/ |
| 18:35.56 | irroot | jkroon true |
| 18:36.19 | irroot | break it till its fixed :P |
| 18:36.22 | Qwell | frem: not really sure what to tell you.. that sounds a whole lot like jitter. |
| 18:36.58 | frem | Qwell: So something like setting up a VPN wouldn't help? |
| 18:37.03 | Qwell | no |
| 18:37.24 | Qwell | in fact, it might hurt |
| 18:37.52 | jkroon | irroot: jkroon - breaking asterisk since October 2007. The end is not yet in sight. |
| 18:38.43 | irroot | frem p2p filtering can screw with voip ... |
| 18:38.50 | *** join/#asterisk Kalavera (~mborda@200.106.87.250) |
| 18:41.14 | pabelanger | Qwell: new blood, quick corrupt Felix1125 |
| 18:41.38 | Qwell | JonathanRose has them! |
| 18:41.42 | frem | Oh, man. "High speed business cable" apparently means "512 Kbs up" *facepalm* |
| 18:41.47 | Qwell | frem: yes. |
| 18:42.09 | Kalavera | hello does someone has asterisk installed on gentoo |
| 18:42.43 | p3nguin | That's ridiculous... I get 5 Mbps up at home. |
| 18:42.57 | Qwell | but does it have "business" in the name? |
| 18:43.03 | frem | Thanks Qwell, leifmadsen :) |
| 18:43.10 | p3nguin | No, of course not. |
| 18:44.17 | jkroon | Kalavera, yes, #gentoo-voip for those of us that use it a lot. |
| 18:45.03 | irroot | fondles Felix1125 i like new blood |
| 18:45.12 | jkroon | *** glibc detected *** /usr/sbin/asterisk: free(): invalid next size (fast): 0x00000000013243b0 *** |
| 18:45.28 | jkroon | irroot, glibc just came up with that. |
| 18:45.29 | *** join/#asterisk Assuero (~Assuero@187.59.194.156) |
| 18:45.36 | Assuero | hi all |
| 18:45.54 | irroot | something not happy |
| 18:46.56 | *** join/#asterisk x1user (~chatzilla@84.40.127.165) |
| 18:47.50 | jkroon | clearly. and the winner is the ast_free(peerarray) at line 16255 (after adjusting for the offset due to extra error checking code - which did NOT trigger) |
| 18:48.27 | x1user | HI i have exten => s,n,Read(gotnum,,30) and when i enter numbers i got from asterisk user entered no numbers :( ? |
| 18:49.13 | irroot | jkroon keep track / make notes |
| 18:50.27 | jkroon | well, i already know it's going through that free - the question is what exactly glibc MEANS with the invalid next size, and secondly - how can I track where it gets corrupted (or whether it's a glibc bug - unlikely as the rest of my systems seems very stable) |
| 18:52.19 | *** join/#asterisk kerx (~kerx@76-240-161-60.lightspeed.irvnca.sbcglobal.net) |
| 18:52.35 | jkroon | and as per experience from a ast_str bug in 1.6.2.18 that I tracked - the issue can really be ANYWHERE ... |
| 18:54.33 | jkroon | as in this case ... getnameinfo |
| 18:55.43 | irroot | indeed it could be arb unrelated issue |
| 18:55.56 | jkroon | ai_family warning, unreleated. |
| 18:57.11 | jkroon | just ran with MALLOC_CHECK_=3 and got this: http://pastebin.com/ji1Picnw |
| 19:04.09 | jkroon | [Sep 20 21:03:49] WARNING[1450]: config.c:1174 process_text_line: No '=' (equal sign) in line 1219 of /etc/asterisk/uls/sip-register.conf |
| 19:04.19 | jkroon | that file doesn't HAVE 1219 lines in it. |
| 19:05.02 | jkroon | in fact, it's an empty file ... |
| 19:05.21 | Qwell | it's included from something that, in total, does. |
| 19:06.15 | jkroon | hmm, actually no, re-occured on the same ast run with a much higer line count - line count probably doesn't get reset. |
| 19:14.38 | Kalavera | jkroon: thank you |
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| 20:21.21 | d_preston215 | Nobody has experience with TDMoE, right? |
| 20:21.54 | leifmadsen | people use tdmoe? |
| 20:23.07 | d_preston215 | I think that answers my question. |
| 20:24.03 | d_preston215 | DAHDI 2.5 locks up my server whenever I try to run dahdi_cfg. |
| 20:24.26 | d_preston215 | And I'm probably the only one in the world that apparantly is still using redFONEs. |
| 20:27.11 | WIMPy | A few months ago someone offered them for free, because he just wanted to get rid of them. |
| 20:28.17 | d_preston215 | They make failover easy though. |
| 20:29.03 | anonymouz666 | why people try to avoid TDMoE? |
| 20:29.09 | anonymouz666 | it does not work well? |
| 20:29.41 | anonymouz666 | d_preston215: that was redphone does |
| 20:30.05 | anonymouz666 | I have never used TDMoE - I don't know if it is reliable |
| 20:30.26 | anonymouz666 | but I would like to hear from the people that have used it |
| 20:30.57 | d_preston215 | Been running redFONE's and on a 1.6 2-server failover setup for the longest. |
| 20:31.38 | d_preston215 | Just that TB is dead and it needs to be migrated to 1.8 at some point anyway. |
| 20:31.49 | WIMPy | I volunteered to take one of them for educational purposes, but it didn't show up. So I've got NFI. |
| 20:33.12 | WIMPy | Many channels took a long time to become available for 1.8 and now 10 is already lingering on the doorstep. |
| 20:33.16 | d_preston215 | Was able to rangle the latest FreePBX into a nice HA-Failover setup. |
| 20:34.12 | d_preston215 | I just don't get why the server takes a dump whenever it tries to apply the dahdi configuration. |
| 20:35.44 | VoipForces | d_preston215: what version of asterisk ? |
| 20:35.49 | d_preston215 | 1.8 |
| 20:36.10 | d_preston215 | I stopped Asterisk before starting dahdi. |
| 20:36.10 | VoipForces | d_preston215: Ok, I saw that in 1.6.2.19, did not have trouble, yet, with 1.8 |
| 20:36.28 | d_preston215 | No problems with 1.6 |
| 20:37.28 | d_preston215 | Been running the last release of TB without DAHDI 2.3.0 crapping out for a good year and a half now. |
| 20:37.58 | d_preston215 | I'm thinking maybe its an interrupt issue somewhere. |
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| 20:38.06 | d_preston215 | Not sure. |
| 20:38.16 | WIMPy | What other dahdi stuff do you have? |
| 20:38.34 | d_preston215 | DAHDI and DAHDI-tools. |
| 20:38.42 | WIMPy | hardware |
| 20:39.06 | d_preston215 | redfone. |
| 20:39.35 | WIMPy | So if you don;t have any dahdi hardware in your PC, how could it possibly be an IRQ issue? |
| 20:39.44 | d_preston215 | I'm using dahdi_dummy to provide timing. |
| 20:40.12 | d_preston215 | Maybe between the RAID card and the ethernet card. |
| 20:40.28 | WIMPy | If there was an issue with dahdi_dummy, I'm sure we would have heard about that. |
| 20:40.38 | d_preston215 | Yeah. |
| 20:40.52 | d_preston215 | It loads the modules correctly. |
| 20:41.14 | d_preston215 | Just that applying the configuration kills the server. |
| 20:45.35 | d_preston215 | I'm thinking about possibly downgrading the version of DAHDI to 2.3.0. |
| 20:45.53 | d_preston215 | Is DAHDI compatible with CentOS 5.5? |
| 20:46.02 | d_preston215 | DAHDI 2.5 I mean. |
| 20:54.34 | ChannelZ | unless CentOS stopped being Linux, it aught be |
| 20:55.24 | *** join/#asterisk luke-jr (~luke-jr@2001:470:5:265:222:4dff:fe50:4c49) |
| 20:55.41 | luke-jr | any idea why Asterisk is compiling AEL to use | delimiters when it doesn't support them? -.- |
| 20:56.57 | leifmadsen | luke-jr: no idea... does asterisk.conf have the appropriate settings to use commas instead of pipes? |
| 20:57.26 | leifmadsen | app_set=1.6 |
| 20:57.31 | luke-jr | leifmadsen: never seen that in, nor touched, asterisk.conf |
| 20:57.35 | leifmadsen | (that means to use commas instead of pipes) |
| 20:57.41 | leifmadsen | perhaps that's the problem |
| 20:57.45 | luke-jr | it's 1.6 |
| 20:57.48 | leifmadsen | beyond that, no idea, I don't use AEL |
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| 21:07.51 | d_preston215 | *Dunce cap* |
| 21:08.40 | d_preston215 | So its not DAHDI itself that is killing the server. |
| 21:08.56 | d_preston215 | pam_console_app takes 100% of cpu whenever I start DAHDI. |
| 21:10.43 | ChannelZ | hmm.. well being drivers is it an access thing? (pam?) |
| 21:11.09 | ChannelZ | SELinux perhaps |
| 21:12.08 | d_preston215 | I thought selinux was disabled. I will look and see. |
| 21:15.15 | d_preston215 | Selinux is disabled. |
| 21:15.35 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:15.45 | ChannelZ | are you starting them as root? |
| 21:16.22 | d_preston215 | Yes. |
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| 21:16.40 | ChannelZ | hmm |
| 21:17.12 | d_preston215 | I halfway want to just disable the damn thing. |
| 21:17.20 | d_preston215 | I want to see if I can fix it though. |
| 21:19.46 | d_preston215 | *Dunce cap again* |
| 21:19.54 | d_preston215 | Its not pam_console_app. |
| 21:20.01 | d_preston215 | It's pam_console_apply |
| 21:20.33 | *** join/#asterisk docid (~PISSS@69.17.142.138) |
| 21:21.08 | docid | ok, i got an odd one yet again.... any call that gets forwaded through my system with no caller ID shows up on the recieving party as a paticular number, same every time |
| 21:21.17 | docid | any clues as to where this is coming from? |
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| 21:21.36 | docid | if it has caller id it gets forwarded properly |
| 21:22.04 | ChannelZ | PSTN call? |
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| 21:23.10 | docid | yes |
| 21:23.26 | docid | both sides pri, using asterisk in a softswitch capacity |
| 21:24.15 | docid | im just wondering if there is a persistant variable or something stuck somewhere causing this |
| 21:25.53 | ChannelZ | well AFAIK caller ID name isn't transmitted, it's looked up from a database. |
| 21:26.08 | docid | right, this is just numbers |
| 21:26.09 | docid | no names |
| 21:26.39 | ChannelZ | oh I misread your question as name not numbers |
| 21:26.50 | ChannelZ | What number does it show up as? |
| 21:27.06 | docid | a valid number we own |
| 21:27.34 | ChannelZ | well depending on your dialplan it could be coming from the configured CID number of your DAHDI channels |
| 21:27.53 | docid | well, ive got the dahdi channels set up as 2 trunks, with no callerid override |
| 21:28.19 | docid | unless i missed it somewhere, but i grepped my whole config directory for the number and its not there |
| 21:28.59 | ChannelZ | It could come from a number of places, including perhaps the telco assigning it if you're actually sending it nothing |
| 21:30.06 | ChannelZ | You said 'any call that gets forwarded thruogh my system' - what is your setup exactly? IE do you mean 'forwarded' as in an incoming call to a local device (SIP?) whose away and forwards their calls to another number, or...? |
| 21:30.07 | docid | aye, were trying to gather enough evidence to force them to investiggate it, they are insisting its not them |
| 21:30.19 | docid | not realy any local devices |
| 21:30.25 | docid | cell system on one side, telco on the other |
| 21:30.37 | docid | pri on either side, as well as iax trunks to other sites |
| 21:31.52 | ChannelZ | Generally the CID will be that of whomever originated the call. If you don't Answer() the incoming channel and then Dial() the outgoing, it should pass along whatever CID the incoming channel had. If you DO Answer first, then Asterisk is technically originating the new outgoing call, and that CID can come from configs |
| 21:32.32 | docid | hrmm, ok, well, i have to answer then dial, or the calling party gets no ringing or anything until it connects |
| 21:33.10 | ChannelZ | So then the question is whether you're actually sending your own number, or sending nothing and the telco is substituting it |
| 21:33.21 | ChannelZ | But it almost doesn't matter because it's not what you want anyway |
| 21:34.18 | docid | well, ive got it echoing out all the caller id info to the console on both call initiation and on hangup, |
| 21:34.38 | docid | and im convinced its being sent from the telco, but i cant prove it |
| 21:34.41 | *** join/#asterisk Jareeta (~ahailes@DR-WILY.MIT.EDU) |
| 21:35.21 | ChannelZ | just for fun if you Set(CALLERID(num)=6666666666) just prior to your Dial, do you get what you set? |
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| 21:36.00 | docid | yes, did that test earlier |
| 21:36.07 | docid | got something else weird |
| 21:36.15 | ChannelZ | but not what you set |
| 21:36.20 | Jareeta | I have a question about GotoIfTime accuracy. Specifically, the wiki says it's accurate to 2 minutes in 1.4. Does this mean it rounds down to the nearest even minute? Or that it may have unspecified behavior in the 2 minute bubble around the time? |
| 21:36.33 | docid | dialplan getting out of order, i have a match for _X. on caller ID, but the fallthrough at the end is what is being parsed |
| 21:36.40 | docid | no, it gets whatever is set |
| 21:36.43 | docid | just when not set |
| 21:37.30 | docid | quick question, if i do a dialplan show <context> is the order its diaplayed in the matching order? |
| 21:38.20 | dr0ck | yes |
| 21:38.44 | docid | ok, goes from longest wildcard to shortest |
| 21:38.45 | docid | grr |
| 21:39.22 | docid | sorry, im rather scattered at the moment |
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| 21:40.35 | Jareeta | hmm. Anyone? |
| 21:40.46 | ChannelZ | I've no personal experience with PRI so I can't make any more guesses besides that your chan_dahdi channels have a number assigned to them, or there's some other macros or something your dialplan is doing which is setting it that you're not seeing. Or the telco is doing whatever they want :) |
| 21:42.39 | docid | they do whatever they want, they have a monopoly |
| 21:43.31 | ChannelZ | Jareeta: I think it essentially "rounds down" |
| 21:44.08 | Jareeta | ok. Good. That means it's at least predictable, even if it is a bit wonky |
| 21:44.21 | ChannelZ | It's like a clock that only changes once every 2 minutes (even), unaware of what happens in between |
| 21:44.56 | luke-jr | reports https://issues.asterisk.org/jira/browse/ASTERISK-18593 |
| 21:45.10 | ChannelZ | docid: not sure if pri debug would reveal anything interesting to you |
| 21:45.41 | Jareeta | nice, thanks so much |
| 21:45.43 | docid | ive tried, its soooo hard to keep up with the scroll with that on |
| 21:46.03 | docid | theres usually several dozen to a hundred calls between the weirdness |
| 21:46.21 | docid | even digging the logs after is tough |
| 21:46.32 | docid | but ive watched it go by and cant find anything of use |
| 21:47.13 | docid | thanks anyhow, if i come up with a mor consise question i cant answer ill be back :) |
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| 22:23.28 | leftist | what is the command to enter asterisk monitor mode? |
| 22:23.44 | navaismo | monitor? |
| 22:23.49 | leftist | sorry console |
| 22:23.52 | navaismo | asterisk -r |
| 22:23.58 | leftist | ahh thanks |
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| 22:48.58 | pdtpatrick1 | Question .. can someone point me a decent configuration for jabber/asterisk integration |
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| 22:59.49 | ChannelZ | for google talk/voice? |
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| 23:19.53 | d_preston215 | I'm just going to assume that DAHDI higher than 2.3.0 doesn't work with redfones. |
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| 23:31.49 | pdtpatrick1 | ChannelZ, for jabber.. i've got it working thus far |
| 23:31.58 | pdtpatrick1 | took some time but figured it out |
| 23:32.03 | pdtpatrick1 | so now i just gotta clean it up |
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| 23:55.52 | pdtpatrick1 | Question .. does anyone know of a bot that will permit you to dial an extension through jabber? |
| 23:56.22 | leifmadsen | pdtpatrick1: you can do that in asterisk 10 I think |
| 23:57.02 | pdtpatrick1 | how about 1.6 or 1.8 ? im not sure i would want my servers running 10 just yet |