| 10:01.35 | *** join/#asterisk infobot (ibot@rikers.org) |
| 10:01.35 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.2.3 (2010/01/26), 1.6.2.16.1 (2010/01/18), 1.4.39.1 (2010/01/18), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
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| 10:18.52 | Vec | schmidts : The other day you sent me a diff that you said I could create a patch from to resolve bug id 18647 (chanspy causing channel hangup). The diff was of pbx.c, is that the only file that requres patching ? |
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| 11:09.44 | DelphiWorld | hello guys |
| 11:09.53 | DelphiWorld | anyone used cisco as5300 with asterisk? |
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| 11:26.41 | schmidts | vec as far as i remember yes |
| 11:33.22 | oblom | When asterisk sends invite to local user for call that arrives from outside (sip) he always rewrites From so it appears like it from local domain ? |
| 11:35.08 | kaldemar | oblom: not quite. it sends a new invite with the internal address though. |
| 11:36.15 | oblom | it's what i meant :) Is it possible to do something so original address will preserved in new invite or it will break something ? |
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| 12:01.18 | Vec | schmidts : when I try to compile the patched version I get undefined reference to `ast_channel_clear_softhangup' |
| 12:01.40 | Vec | What kind of stability could I expect from Asterisk 1.4 SVN ? |
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| 14:09.35 | Katty | morning |
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| 14:47.36 | Amnesia | is there any good step by step guide or w/e for asterisks configuration? |
| 14:47.54 | Amnesia | I'm editing extensions.conf atm but it still kinda looks like gibberish |
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| 14:56.18 | oblom | Amnesia: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html/asterisk-book.html#asterisk-CHP-5-SECT-1 |
| 14:56.47 | Amnesia | oblom: thx $$ |
| 14:56.59 | cd101 | hello all. I am new to Asterisk. I am looking for clarification. Do I need to rebuild asterisk to get support for new codecs/formats? I am looking to integrate a new codec into 1.4->current |
| 14:58.26 | Faustov | hah |
| 14:58.31 | Faustov | I didn't know there's a second edition |
| 14:58.51 | WIMPy | Or a third? |
| 14:58.53 | Faustov | oblom: any idea if it was recently released? |
| 14:59.22 | WIMPy | http://ofps.oreilly.com/titles/9780596517342/ |
| 14:59.24 | oblom | no clue |
| 15:02.28 | pabelanger | cd101: which codecs/formats? |
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| 15:04.59 | cd101 | It uses a custom format |
| 15:07.04 | pabelanger | cd101: You should look at https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal |
| 15:07.10 | cd101 | it is definitly not listed in format.h, which is why I was asking. It appears that I would have to rebuild a custom Asterisk with support for a newly defined format. |
| 15:07.18 | cd101 | thank you. Looking.. |
| 15:08.28 | pabelanger | cd101: dvossel is in the middle of the rewrite. He is also adding new codecs now (SILK, more wideband support, etc) |
| 15:10.30 | WIMPy | Is there BTW any info on the progress on that? |
| 15:12.13 | WIMPy | Or on the target? Will it be complete for 1.8.4? |
| 15:12.28 | pabelanger | WIMPy: dvossel has been giving updates on the weekly asterisk developers conference |
| 15:12.32 | pabelanger | no, 1.10 |
| 15:12.37 | WIMPy | Or do we get some intermediate state(s)? |
| 15:13.58 | WIMPy | just noticed that the old stuff doesn't work any more, but the new features as per the web page don't either. |
| 15:14.10 | schmidts | wimpy dvossel is know working on the fixtheworld phase 2 and AFAIK there are 3 phases but i think they will only come for 1.10 not for 1.8 |
| 15:14.27 | malcolmd | correct, only for 1.10 |
| 15:15.24 | kaii | is there any active effort to support RTAudio in asterisk? |
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| 15:16.39 | malcolmd | kaii: no, digium is making no efforts to license rtaudio from microsoft. we haven't been aware of any substantial market interest in such a thing that would encourage us to pursue that avenue. it's not an open codec, so... |
| 15:17.00 | kaii | ok, thx for the clear answer. |
| 15:18.02 | malcolmd | np. if people start beating down our doors for it and it looks like we can recoup our engineering and legal expenses, it's something we'd consider. |
| 15:18.58 | malcolmd | what's your interest? just that it's used by lync / ocs? is there a technical advantage to it over other codecs? |
| 15:19.02 | kaii | was just wondering if there would be any chance to integrate asterisk with OCS natively without needing a media gateway, now that asterisk supports TCP and MTLS.. |
| 15:19.11 | malcolmd | ah |
| 15:19.14 | kaii | s/media gateway/"mediation server"/ |
| 15:20.44 | WIMPy | BTW: Does anyone know the current legal status of the GSM HR and EFR CODECS? I've seen that they are supported by OpenBSC. |
| 15:22.09 | malcolmd | wikipedia indicates that EFR is covered by patented ACELP technology and licensed by VoiceAge |
| 15:22.56 | WIMPy | I read that, but I couldn't find any info on the age of that patent. |
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| 15:24.38 | pabelanger | Good ole microsoft and proprietary software |
| 15:25.05 | malcolmd | for 1.10, we've got a complete set of slinear modes now, and the first two codecs to take advantage will be speex32kHz and silk - available at 8, 12, 16, and 24kHz |
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| 15:26.55 | wonderworld | is there a more simple voicemail app for asterisk? basicly i'd just need listen to messages, jump to next/prev and delete. Everything without long anouncements. |
| 15:27.47 | pabelanger | wonderworld: app_minivm |
| 15:27.58 | wonderworld | thanks, i'll have a look |
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| 15:37.45 | Amnesia | hm, how can I change my voicemail lang to nl instead of the default english voicemail? |
| 15:37.55 | Amnesia | already placed the folder to /var/lib/asterisk/sounds/ |
| 15:38.32 | WIMPy | It uses the language of the channel, as does everything else. |
| 15:38.47 | Amnesia | hm, and how can I change it? |
| 15:38.55 | Amnesia | the channel language:) |
| 15:39.54 | WIMPy | Either in the channel/peer config or in the dialplan. |
| 15:40.11 | WIMPy | CHANNEL(language) |
| 15:40.40 | Amnesia | sorry I'm still kinda new with asterisk |
| 15:40.47 | Amnesia | but in which configfile |
| 15:41.10 | Amnesia | and I guess it'll also not understand: "CHANNEL(nl)" on one single linge |
| 15:41.13 | Amnesia | line* |
| 15:41.15 | WIMPy | You didn't say how you ae calling, so I can't be more specific. |
| 15:41.29 | Amnesia | through a sip phone |
| 15:41.37 | WIMPy | No, Set(CHANNEL(language)=nl) |
| 15:41.44 | WIMPy | Then you can change it in sip.conf. |
| 15:42.27 | Amnesia | at the top of http://pastie.org/1562959 ? |
| 15:42.55 | wdoekes2 | Amnesia: in the extensions.conf |
| 15:42.57 | Amnesia | is that correct? |
| 15:43.00 | Amnesia | ah |
| 15:43.10 | wdoekes2 | before the VoiceMail() |
| 15:43.37 | Amnesia | there is not VoiceMail() in my extenstions.conf^^ |
| 15:43.37 | WIMPy | in sip.conf it's just language=nl. |
| 15:43.56 | WIMPy | VoiceMailMain(... |
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| 15:44.21 | Amnesia | woot it's working |
| 15:44.56 | Amnesia | thx:D |
| 15:45.05 | Amnesia | language=nl apparantly already worked:p |
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| 16:26.42 | Amnesia | hm, how can I change the default dahdi context? |
| 16:27.07 | Amnesia | /etc/asterisk/chan_dahdi.conf -> context=foobar? |
| 16:27.10 | russellb | change the "context" option in /etc/asterisk/chan_dahdi.conf |
| 16:27.12 | russellb | yes. |
| 16:27.24 | *** join/#asterisk jm|laptop (~jamiem@irc.jamiem.com) |
| 16:27.29 | jm|laptop | hello :) |
| 16:27.49 | Amnesia | russellb: already did that |
| 16:27.56 | Amnesia | it still states default |
| 16:28.08 | russellb | make sure you put it _before_ the channel => line(s) |
| 16:28.43 | Amnesia | k sec:) |
| 16:28.45 | Faustov | hah, I struggled with that on the days |
| 16:29.02 | russellb | chan_dahdi.conf is weird |
| 16:29.06 | jm|laptop | I'm using cdr_mysql and I'm trying to work out what to enter in the config file so that enum translated numbers are recorded |
| 16:29.15 | jm|laptop | is there an alias I should use? |
| 16:29.50 | Amnesia | ^ |
| 16:30.10 | jm|laptop | lastdata is showing me the translated SIP/IAX address, dst is showing ~ sw-150-sip |
| 16:30.37 | Amnesia | when I enter : "context=outgoing" at the top of /etc/asterisk/chan_dahdi.conf there arent any channels at all:/ |
| 16:30.55 | Amnesia | (outgoing does exist in my extensions.conf) |
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| 16:34.13 | Amnesia | goddamnit |
| 16:35.08 | Amnesia | now I've typed "context=outgoing" underneath [channels] it's set to default again:S |
| 16:36.30 | Amnesia | Faustov: how'd you fix it? |
| 16:43.22 | Faustov | ffs, don't ask questions then disappear ;) |
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| 16:43.55 | leifmadsen | indeed |
| 16:47.00 | jaytee | patience is a virtue....yet there are so few virtuous people left. |
| 16:48.07 | tzafrir | Faustov, he forgot he asked them |
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| 16:59.25 | bmoraca_work | how is Asterisk's MGCP support? is it decent? |
| 17:01.58 | Qwell | bmoraca_work: not really |
| 17:02.15 | bmoraca_work | hrm |
| 17:03.23 | bmoraca_work | match_auth_username doesn't work for peers anymore, so i'm looking for a better way to get my Adtran TA900 FXS ports to work with Asterisk |
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| 17:18.46 | Kobaz | bmoraca_work: what i do is have to block the adtran ip on all my local sip users, and allow it on the fxs users... and i match on the station number defined in the adtran |
| 17:18.46 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 17:19.31 | Kobaz | otherwise if i get a callerid from the adtran like say 5000, which also matches a sip user 5000, asterisk matches the wrong user and the call is rejected... so i have to deny the adtran ip all over the place |
| 17:19.40 | Kobaz | i wish they supported a from_user type thing |
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| 17:31.18 | bmoraca_work | Kobaz: the problem is that i've got multiple FXS ports on my Adtrans. so, the first one to register is the only one that ends up being usable...the rest of them come back with a username mismatch |
| 17:31.29 | Kobaz | let me see what i do |
| 17:32.09 | bmoraca_work | in a 1.6.2.0, match_auth_username took care of it without issue |
| 17:32.23 | bmoraca_work | on 1.6.2.10 and 1.6.2.16.1, match_auth_username doesn't seem to do anything |
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| 17:32.57 | bmoraca_work | problem is that 1.6.2.0 has bugs in t.38 passthru, so I can't use it anymore |
| 17:33.12 | Kobaz | oooh |
| 17:33.18 | Kobaz | you kno |
| 17:33.27 | Kobaz | i never set up an adtran fxs with asterisk |
| 17:33.33 | Kobaz | it just comes in on the t1 and goes straight to the fxs |
| 17:33.36 | Kobaz | they are fax machines |
| 17:33.44 | bmoraca_work | ahh |
| 17:33.55 | jkroon | bmoraca_work, are those issues resolved in 1.6.2.16 already or not? |
| 17:34.31 | bmoraca_work | jkroon: match_auth_username doesn't work worth shit in 1.6.2.16.1. the t.38 bugs I was seeing have not appeared in 1.6.2.10 or 1.6.2.16.1 |
| 17:34.47 | brainiac | does anyone know why a call would have terrible audio quality problems when you barge in on it via a graphic manager? |
| 17:35.04 | jkroon | bmoraca_work, match_auth_username ? |
| 17:35.17 | jkroon | good to know t.38 is supposed to work finally in asterisk. |
| 17:35.46 | bmoraca_work | it's supposed to tell Asterisk to match a peer based on the username provided in the proxy-authentication or the digest authentication, rather than use From: header or IP address |
| 17:36.06 | Kobaz | brainiac: that makes no sense.. the tool you use to send commands is going to have nothing to do with how asterisk handles the call |
| 17:37.27 | brainiac | regardless of the tool, a call, when barged in on sounds VERY distorted. |
| 17:37.39 | Kobaz | how are you doing a barge |
| 17:38.06 | bmoraca_work | sounds like it could be a timing issue |
| 17:38.29 | brainiac | Kobaz: using a tool called iview |
| 17:38.38 | Kobaz | that doesn't tell me anything |
| 17:38.43 | Kobaz | what command is it sending to the manager |
| 17:39.17 | brainiac | that's what I'm in the process of finding out |
| 17:39.58 | Kobaz | tcpdump -i <interface> port 5038 -s0 -A |
| 17:41.10 | brainiac | thank you. brb |
| 17:42.20 | bmoraca_work | iview creates a meetme room and transfers all channels into that meetme room |
| 17:43.33 | jkroon | older versions of meetme had to have a dahdi timing source. |
| 17:43.38 | brainiac | there's noone in the office where the phone system is located, so I'll try again later |
| 17:45.00 | brainiac | jkroon: I should be able to fix that |
| 17:46.03 | Kobaz | run dahdi_test |
| 17:47.41 | Katty | mmmsammich |
| 17:51.56 | jkroon | brainiac, do you know if the ChanSpy() application has the same problem? |
| 17:53.15 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 17:53.56 | brainiac | I don't know, how can I test that? |
| 17:54.12 | Kobaz | dahdi_test |
| 17:54.33 | Kobaz | chanspy doesn't use dahdi so it would be fine, but run the timing test |
| 17:54.52 | Kobaz | crpapy audio in meetme is a dahdi problem more likly than not |
| 17:56.12 | brainiac | ok |
| 18:08.47 | jkroon | brainiac, set up an extension to go into ChanSpy(SIP/foo) where foo is the channel name of some sip channel, eg ChanSpy(SIP/100,ds) will spy on a SIP channel starting with 100. |
| 18:09.23 | jkroon | Kobaz, never versions of asterisk uses timerfd right? |
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| 18:25.52 | leifmadsen | jkroon: newer versions *can* use timerfd |
| 18:26.18 | leifmadsen | (they can also use res_timing_dahdi, res_timing_pthread, and res_timing_timerfd) |
| 18:26.26 | leifmadsen | I still prefer res_timing_dahdi though |
| 18:27.20 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
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| 18:38.10 | jkroon | leifmadsen, any particular reason why? |
| 18:41.55 | brainiac | jkroon: ChanSpy() is the app that is having the prob I mentioned |
| 18:42.10 | Kobaz | leifmadsen: why is dahdi a better option? |
| 18:42.42 | russellb | over timerfd? |
| 18:43.14 | russellb | res_timing_pthread is just inefficient and eats CPU |
| 18:43.19 | russellb | dahdi and timerfd both seem to work fine |
| 18:43.39 | russellb | timerfd is nice in that it doesn't require you to install something extra (DAHDI) if you didn't need it otherwise |
| 18:43.54 | *** join/#asterisk tyman_ (~tyler@173-12-219-179-Fresno.hfc.comcastbusiness.net) |
| 18:44.00 | Kobaz | yeah |
| 18:44.15 | Kobaz | i was thinking timerfd is probably the best bet if you dont have dahdi hardware |
| 18:44.23 | russellb | yup |
| 18:44.43 | drmessano-lt | Shame that CentOS 5's kernel is too old |
| 18:45.00 | russellb | you have the power to update it |
| 18:45.06 | drmessano-lt | Of course |
| 18:45.31 | russellb | but it does take effort |
| 18:45.35 | drmessano-lt | But then you void the GPL warranty and the RPM ninjas come get you |
| 18:45.35 | russellb | and that's no fun |
| 18:46.43 | drmessano-lt | Ok no, but it's not like they have some extended repo with a newer kernel, so for most it would become another administrative burden |
| 18:47.09 | drmessano-lt | But after 6.x gets out there, it will be nice to just use timerfd |
| 18:50.20 | *** part/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 18:51.28 | drmessano-lt | Is the 1.8 branch known to currently be broken in SVN? |
| 18:52.18 | drmessano-lt | I updated last week and had to roll back like 2 weeks to get a working build. Updated last night and the same. |
| 18:53.18 | russellb | broken in what way? |
| 18:54.42 | drmessano-lt | It compiles, can register devices to it.. but no calls complete. I havent dug more into it because I rolled back to working revisions and rebuilt both times |
| 18:54.50 | drmessano-lt | SIP <> SIP |
| 18:57.50 | carrar | Who uses SIP anyways!! |
| 18:58.21 | WIMPy | Everyone how can;t afford something decent. |
| 18:59.52 | tzafrir | drmessano, a RHEL5-compatible with a newer kernel: http://www.oracle.com/us/technologies/linux/index.html |
| 19:00.37 | drmessano-lt | ooooh |
| 19:00.53 | tzafrir | But then again, this is Oracle |
| 19:01.16 | carrar | UNBREAKABLE!! |
| 19:01.32 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 19:01.33 | tzafrir | (because it's already broken) |
| 19:01.38 | drmessano-lt | lol |
| 19:01.43 | *** part/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 19:08.34 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 19:27.04 | *** join/#asterisk Besticles (~larry@209-58-227-178.static-ip.telepacific.net) |
| 19:29.00 | Besticles | Question. I'm writing a ChanSpy piece, plugged in AMI support to detect who chanspy is spying on. I wanted to use that AMI to PlayDTMF to control chanspy within my app. PlayDTMF gets queued successfully, and you can hear it on the chanspy. But it doesn't do what it's suppose to do (Move to the next agent). Any suggestions? |
| 19:29.26 | *** join/#asterisk teloniusz (goldie@inferno.hell.pl) |
| 19:30.12 | teloniusz | I'm trying to dialout with an analog card, but I get: [Feb 14 20:28:54] WARNING[2560]: app_dial.c:1547 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 19:30.21 | teloniusz | how to debug this more closely? |
| 19:31.43 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 19:31.45 | WIMPy | Besticles: You can't use playdtmp as a remote control fro dialplan applications. |
| 19:32.23 | WIMPy | teloniusz: Looks like you don't have chan_dahdi loaded. Possibly because it's not configured properly. |
| 19:33.42 | WIMPy | Besticles: How are you playing dtmf at all, while you're in chanspy? |
| 19:33.50 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 19:35.08 | Besticles | WIMPy: I was using the PlayDTMF in Manager Interface. |
| 19:35.23 | Besticles | It definitely plays it, just doesnt remote control it. |
| 19:35.47 | WIMPy | Sure. It plays it to the user, not the other way round. |
| 19:35.52 | Besticles | I figured I was doing something wrong since the AMI event traffic supports me being able to detect exactly who we are spying on. |
| 19:35.53 | teloniusz | WIMPy: I'd like it to be so simple, but 'dahdi show channels' tells me that there are 16 channels, all "In Service" |
| 19:36.05 | Besticles | So, I need to use the API for the actual soft phone to get this done. |
| 19:36.08 | Besticles | Gotchya, thanks. |
| 19:37.21 | WIMPy | teloniusz: Ok, so did you specify a valid channel or group then? |
| 19:38.17 | drudge` | any sip resellers to recomend? currently using dash carrier services |
| 19:42.10 | teloniusz | WIMPy: how do I check that? There's a single phone line connected to first socket of my card |
| 19:42.30 | teloniusz | but waaait, maybe I should have used a splitter... |
| 19:42.42 | WIMPy | What does your Dial look like? |
| 19:46.42 | teloniusz | WIMPy: ok, disregard this; I'm just quite stupid |
| 19:47.18 | teloniusz | I've connected RJ-11 2-wire plug to RJ-45 8-wire socket |
| 19:47.39 | teloniusz | now it works :> |
| 19:48.55 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 19:53.11 | *** join/#asterisk dacm_work (~dan@host86-166-170-67.range86-166.btcentralplus.com) |
| 19:53.15 | dacm_work | HI guys. |
| 19:54.26 | dacm_work | Often it seems that when I receive a call, they do not hear me immediately and are normally the first to say `Hello?'. Is this likely a firewall issue? (i.e. They must speak first before the connection is made.) |
| 19:54.49 | *** join/#asterisk provolone (~root@c-68-33-201-70.hsd1.md.comcast.net) |
| 19:54.51 | dacm_work | I'm running asterisk behind a NAT, but the required ports _should_ be forwarded. |
| 19:55.03 | dacm_work | Trying to track down the problem. |
| 19:55.36 | dacm_work | I also can't divert incoming calls back outside to the PTSN. |
| 19:55.55 | dacm_work | Or rather, I get no audio when doing so. |
| 19:56.19 | dacm_work | Any hints / pointers would be hugely appreciated. |
| 19:57.33 | *** join/#asterisk OneNarrowWay (~OneNarrow@ip4da1344b.direct-adsl.nl) |
| 19:58.14 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 19:59.45 | *** join/#asterisk OneNarrowWay (~OneNarrow@ip4da1344b.direct-adsl.nl) |
| 19:59.46 | provolone | I want to repeatedly call this person who owes me money from multiple numbers until he answers or gets a new phone number. If possible I would like to dial my cell and be connected in the even that he answers. I dont want to use large packages that need lots of config, but just a simple script. Is AGI the simplest solution to my problem? Should I investigate a simpler package than asterisk ? |
| 20:01.52 | provolone | 'in the event' |
| 20:03.48 | *** join/#asterisk xheliox (jeff@pdpc/supporter/student/xheliox) |
| 20:05.39 | *** join/#asterisk Docfxit (~none@netblock-75-79-6-149.dslextreme.com) |
| 20:06.26 | *** part/#asterisk Docfxit (~none@netblock-75-79-6-149.dslextreme.com) |
| 20:07.57 | Kobaz | provolone: make sure you don't violate the fair debt collection practices act |
| 20:08.47 | Kobaz | provolone: get a good lawyer |
| 20:08.50 | provolone | no kneecapping ? |
| 20:11.31 | Qwell | ... |
| 20:14.36 | _Corey_ | lol |
| 20:15.09 | pabelanger | ~sipnat |
| 20:15.09 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 20:15.16 | pabelanger | dacm_work: ^^ |
| 20:15.53 | Kobaz | The FDCPA is a strict liability law, which means that a consumer need not prove actual damages in order to claim statutory damages of up to $1,000 plus reasonable attorney fees if a debt collector is proven to have violated the FDCPA.[25 |
| 20:16.17 | dacm_work | pabelanger: But does it sound like a NAT issue? |
| 20:16.52 | voxter | Anyone know how asterisk 1.4.x decides matching order on incoming call peers that may have the same host= specified? |
| 20:17.07 | pabelanger | dacm_work: Yes, review your settings. |
| 20:17.23 | pabelanger | Could also just be delay between the PSTN network and your ITSP |
| 20:19.35 | _Corey_ | anyone played around with the CONNECTEDLINE function? I'm trying to figure out if it's possible to update the info on a ringing extension or if it needs to be answered. |
| 20:20.25 | _Corey_ | works pretty well otherwise |
| 20:24.20 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 20:25.23 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 20:27.59 | *** join/#asterisk gerhard7 (~gerhard7@212-123-146-122.ip.telfort.nl) |
| 20:33.58 | dacm_work | pabelanger: Ok thanks. |
| 20:34.11 | raden_work | <PROTECTED> |
| 20:34.23 | raden_work | that keeps coming up like every 60 seconds |
| 20:35.27 | raden_work | how often does it need to do a freaking lookup ? |
| 20:35.40 | raden_work | driving me nuts |
| 20:37.10 | *** join/#asterisk JonathanRose (~jonathanR@nat/digium/x-oyxdhwkmfmeuoraf) |
| 20:43.35 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 20:55.51 | pabelanger | raden_work: Then change the setting |
| 20:56.02 | raden_work | what setting is it ? |
| 20:56.44 | pabelanger | raden_work: dnsmgr.conf |
| 20:58.17 | *** join/#asterisk psilikon (~joel@cerberus.vicimarketing.com) |
| 20:58.23 | raden_work | see if that works |
| 21:00.04 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 21:03.05 | *** join/#asterisk WindBack (~quassel@kirk.capitalinasdc.com) |
| 21:04.46 | WindBack | Hi, In which way can I launch an script each time I start asterisk and when I restart it from CLI> doing core restart now |
| 21:04.48 | WindBack | ? |
| 21:09.53 | jkroon | does anybody know whether grandstream also has a nice scheme to allocate mac blocks to phone models like snom does? Eg http://wiki.snom.com/Settings/mac |
| 21:10.02 | jkroon | if so - any ideas where I can obtain it from? |
| 21:10.24 | Qwell | "grandstream" and "nice" don't often go in the same sentence |
| 21:10.53 | jkroon | Qwell, hehe, no, i guess not, but the question stands. |
| 21:11.21 | _Corey_ | It helps to speak Chinese |
| 21:11.48 | jkroon | that would be for the horde of huawei phones branded under various names like ip touch etc ... |
| 21:14.00 | *** join/#asterisk Kyosh (whoa@pool-74-108-17-185.nycmny.fios.verizon.net) |
| 21:15.26 | _Sam-- | im the sole grandstream dissenter.....but ive had nothing but fine service from many old Grandstream GXP2000s here at my place. |
| 21:18.12 | jkroon | got one myself, and a bunch of gxp2020 and 285s and the majority of them work reasonably well, nothing compared to SNOM or Polycom but alas, dirt cheap in comparison and therefor preferred by the majority of my clients :( |
| 21:19.10 | *** join/#asterisk b0ot (~Jinxed---@147.177.57.8) |
| 21:19.36 | Kobaz | i tried out the grandstream "enterprise" fxo gateways, since they are like half the price of anything else |
| 21:19.40 | Kobaz | boy do they suck hard |
| 21:19.43 | b0ot | Is asterisk the only pbx that can do sip registerations without a password? |
| 21:19.48 | b0ot | as opposed to something like cme? |
| 21:19.52 | jkroon | Kobaz, i won't touch one of those again either. |
| 21:20.06 | Kobaz | can't complete calls 50% of the time |
| 21:20.20 | Kobaz | other than that they work fine |
| 21:20.28 | jkroon | if it even picks up the fact that it's ringing or doesn't echo the living crap out of you. |
| 21:20.47 | Kobaz | and tech support takes a month to respond |
| 21:21.04 | Kobaz | i just heard back on friday finally after a month of silence on my issue... "sorry our ticket system accidentally closed your ticket" |
| 21:21.05 | jkroon | To GS's defence they did take some effort to fix it for my case after kicking up quite a stink with my supplier. |
| 21:21.16 | Kobaz | like hell it did... status showed "open" the whole time |
| 21:21.52 | jkroon | oh well, any other ideas on obtaining the model of a detected phone on the network short of telnetting into it? |
| 21:21.54 | Nugget | telnet is eeeeeeevil! |
| 21:22.28 | jkroon | Nugget, if it's the only way I'll get the model (compared to having to ask the user) I'll do it. Will probably use nc rather though. |
| 21:23.26 | *** join/#asterisk batfastad (~benbradle@87-194-37-131.bethere.co.uk) |
| 21:27.10 | jkroon | sleep 0.1 | telnet 192.168.47.55 2>/dev/null | sed -nre 's/^Grandstream ([^ ]+) .*/\1/p' |
| 21:27.25 | jkroon | very unclean but it seems to work against at least GXP2000, GXP285 and BT200 |
| 21:29.03 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 21:29.13 | _Corey_ | jkroon: You could figure out the manufacturer based on the MAC probably |
| 21:29.26 | _Corey_ | jkroon: Wouldn't help with models though |
| 21:30.24 | jkroon | _Corey_, yea, grandstream has the 3-octet prefix 000b82, and SNOM 000413, for SNOM I can then subclass based on the rest of the MAC, but GS doesn't seem to follow any standard. |
| 21:31.06 | jkroon | Those are primarily the two I'm concerned with at the moment, but will probably want to try and add linksys, siemens and cisco later on. |
| 21:31.20 | drmessano-lt | GS uses what MACs they get from the recycled Barbie and 3COM NIC's they build their phones out of.. |
| 21:31.38 | provolone | these configs are so much fun |
| 21:31.47 | drmessano-lt | Dont be surprised to find the door handle from an old Datsun on a Grandstream phone |
| 21:32.17 | jkroon | drmessano-lt, so what phones do you use/prefer? |
| 21:32.47 | drmessano-lt | Well, since Cisco killed off the nice Linksys phones I liked, though they weren't perfect, I prefer Polycom |
| 21:33.23 | jkroon | drmessano-lt, ok, polycom are really cool but I have exactly zero clients willing to fork out for those. |
| 21:33.36 | drmessano-lt | $100 for a phone is too much? |
| 21:33.39 | Qwell | They aren't willing to pay $100? |
| 21:33.56 | jkroon | which polycom phone comes with that price tag? |
| 21:34.06 | _Corey_ | They still have those linksys phones... they're just rebranded cisco now |
| 21:34.11 | drmessano-lt | IP331 for one.. |
| 21:34.14 | jkroon | I sit at the southern most tip of Africa ... |
| 21:34.17 | _Corey_ | Still have SPA model numbers |
| 21:34.23 | Qwell | jkroon: 320s are like $80 |
| 21:34.25 | drmessano-lt | _Corey_, they are not the same under the hood anymore |
| 21:34.44 | jkroon | Qwell, not quite over here. |
| 21:34.47 | drmessano-lt | _Corey_, I deployed a few.. they're horribly buggy |
| 21:34.50 | _Corey_ | drmessano-lt: The provisioning template is almost identical |
| 21:34.56 | jkroon | the 320s are more like $120 USD doing a direct translation. |
| 21:35.05 | _Corey_ | I'm not a fan but we had to work with some recently and didn't have many problems |
| 21:35.56 | jkroon | compared to GS which is around $90 doing a direct conversion. |
| 21:36.04 | drmessano-lt | But GS is cheap crap |
| 21:36.21 | drmessano-lt | I can get a GS here in the US for $70.. but why |
| 21:36.37 | jkroon | not argueing. but unfortunately people here think with their wallets not their brains. |
| 21:37.05 | Qwell | So then show them why they'd be throwing money away by using GS. |
| 21:37.06 | jkroon | a crap product at $80 compared to a good product at $90 will innevitably sell in MUCH greater volumes. |
| 21:37.29 | jkroon | Qwell, do tell ? |
| 21:37.43 | Qwell | How much are you spending, in your time, on just this one issue? |
| 21:38.27 | jkroon | only been looking at it for about an hour or two now, and i only need to solve it once. |
| 21:38.39 | Qwell | Now multiply that by 78,000, for the other issues you'll see. |
| 21:38.47 | jkroon | :) |
| 21:38.52 | raden_work | i added another g729 codec file how do i get it to reload |
| 21:39.10 | drmessano-lt | jkroon, that's an argument that holds little water with me. Everyone wants the best value for their money, which begins with "I want whatever is cheapest"... that goes for any product.. But there is reason they sell not only the $250 crap 32-inch TV, but the higher quality 32-inch TV for $300. |
| 21:39.10 | jkroon | asterisk -rx "core restart ???" (probably now) |
| 21:40.16 | jkroon | drmessano-lt, because those who care about a 32" TV is the guy that is going to use it. a typical managerial type gives not a crap. |
| 21:40.33 | jkroon | _typical_ - at least from my experience. |
| 21:40.40 | jkroon | not all. |
| 21:41.05 | jkroon | fortunately. |
| 21:41.12 | drmessano-lt | Well, what are they going to use with these phones, since a PBX is probably more than they need anyone, and should be using POTS phones? |
| 21:41.18 | drmessano-lt | anyway* |
| 21:41.46 | drmessano-lt | Couldnt they just get by with some $20 multiline POTS phones? |
| 21:43.12 | jkroon | they often do. |
| 21:43.34 | jkroon | or rent an analog pabx that basically does incoming+outgoing calls. |
| 21:44.00 | jkroon | nothing interesting. |
| 21:46.39 | jkroon | anyway. SNOM are at least starting to take over slowly but surely at most of my clients. |
| 21:47.13 | jkroon | it takes a few dead phones before they "get it", or "muffled" sound, or some other similar crap. even though you warn them beforehand. |
| 21:48.13 | *** join/#asterisk lanning (~lanning@208.87.233.137) |
| 21:48.56 | *** join/#asterisk tyman_ (~tyler@173-12-219-178-Fresno.hfc.comcastbusiness.net) |
| 21:50.07 | dacm_work | pabelanger: I've made those changes to sip.conf, but it still doesn't seem to work. Is there any way to debug? |
| 21:55.04 | jkroon | ok well guys, thanks for the GS bashing session, i'm off to bed. |
| 21:55.22 | batfastad | Hi everyone. I've been looking for some info on e164.arpa and enum. I'm in the UK and the registry of 4.4.e164.arpa has been delegated to Nominet, the body that operate the .uk domain registry. They list only 2 registrars of uk enum domains, and one of those isn't currently processing applications in the UK zone |
| 21:57.04 | WindBack | exit |
| 21:58.14 | batfastad | But I've found e164.org and that seems like a great idea... what's the status of that? Do most Voip providers list their ranges on it? Guess I should list our numbers as the more on there the better. |
| 22:02.09 | *** join/#asterisk oblom (~oblom@bzq-79-178-198-250.red.bezeqint.net) |
| 22:02.40 | drmessano-lt | I was setup on e164.org but I think something is broken on their end |
| 22:03.06 | drmessano-lt | I couldnt add any new numbers.. the test calls didnt come in and the page to schedule them in broken as well |
| 22:03.28 | drmessano-lt | Feels a little abandoned |
| 22:03.38 | WIMPy | Worked for me, but noone used it so far. |
| 22:04.47 | *** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 22:07.52 | batfastad | Yeah. There was a link for a forum on there and thought I'd look to see how many posts etc were on there but the forum's down. Shame as it's a great idea. |
| 22:10.37 | batfastad | Not surprised e164.arpa's not taking off though as it's not in the telcos interests |
| 22:11.00 | *** join/#asterisk Poincare (~jefffnode@2001:5c0:150f:1704:230:48ff:fe86:1622) |
| 22:13.56 | drmessano-lt | WIMPy, have you tried to add a number lately? |
| 22:15.33 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 22:21.10 | WIMPy | I'd have to donate to do so. |
| 22:33.16 | *** join/#asterisk nsgn (~nsgn@rrcs-67-78-117-241.sw.biz.rr.com) |
| 22:33.37 | nsgn | so i'm getting bruteforced hard core..what's the preferred way to protect a system that must be open to the net on 5060? |
| 22:34.01 | nsgn | our strong passwords are keeping them out, but the number of open channels as they attempt to register is really bogging my bandwidth |
| 22:34.06 | adeel | iptables |
| 22:34.35 | nsgn | is there a way to configure this where after x number of failed sip registrations you get on the blacklist? |
| 22:34.51 | nsgn | because our equipment will never fail this test and i'd love for it to be pretty quick to ban |
| 22:34.58 | leifmadsen | batfastad: see what I'm about to link you for more information about e164.* |
| 22:35.00 | leifmadsen | ~newbook |
| 22:35.00 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 22:35.26 | chazzam | nsgn: look into fail2ban |
| 22:35.31 | WIMPy | nsgn: Listen on AMI and set iptables. |
| 22:35.47 | russellb | the security chapter in that book covers how to set up fail2ban |
| 22:36.17 | nsgn | ok |
| 22:36.18 | nsgn | thx |
| 22:36.46 | *** join/#asterisk MemX (~memx@ab01a.memxit.com) |
| 22:37.23 | MemX | Can someone point me in the right direction for the most 'correct' way to re-compile DAHDI to include OSLEC on AsteriskNOW 1.71 ? |
| 22:38.31 | dacm_work | Hmm, is there any reason why a softphone running on my * box would not be able to send out audio. (Calls connect ok.) |
| 22:38.50 | dacm_work | Is it the same issue as a NAT? |
| 22:44.20 | *** join/#asterisk dailylinux (~fedora@34a744fb09f3b315c220246905467b77.cust.nwn.no) |
| 22:51.51 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 22:52.06 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 22:53.05 | *** part/#asterisk batfastad (~benbradle@87-194-37-131.bethere.co.uk) |
| 22:53.38 | WIMPy | Hmm. No mention of RTP attacks in the security chapter. |
| 22:55.49 | russellb | there are a lot of things we couldn't cover. The realm of what you could potentially cover is so massive .... |
| 22:56.12 | russellb | we felt the SIP security (scanning for accounts and brute forcing passwords) were the most prevalent issues being faced |
| 22:56.32 | russellb | it's certainly not intended to cover all things VoIP security |
| 22:57.13 | *** join/#asterisk exothermc (~miles@74.85.89.146) |
| 22:57.37 | WIMPy | But RTP attacks seem to be a good way to make some cash with you 0900 number from standard Asterisk installations. |
| 22:58.51 | *** join/#asterisk stephenw (~stephenw@stephen-smells.ssimicro.com) |
| 22:59.01 | stephenw | I've got a very simple asterisk configuration. |
| 22:59.10 | stephenw | I'm trying to get a conference bridge working, using ConfBridge |
| 22:59.35 | stephenw | I've got everything apparently working (I can call into the bridge, the logs show attempts to play on-enter sounds) |
| 22:59.42 | stephenw | But asterisk isn't sending out any RTP. |
| 22:59.58 | stephenw | I've verified with tcpdump that it is receiving RTP from the UA |
| 23:00.04 | stephenw | But it isn't sending out any RTP packets. |
| 23:00.09 | stephenw | Any ideas what could cause this? |
| 23:00.39 | toresbe | stephenw: enable RTP debug on the console? |
| 23:00.55 | toresbe | stephenw: rtp set debug on |
| 23:01.06 | stephenw | Yep, did that. |
| 23:01.19 | stephenw | Tonnes of: Got RTP packet from 64.247.129.37:49860 (type 18, seq 023260, ts 3076334795, len 000020) |
| 23:01.28 | stephenw | No indication of sending RTP packets. |
| 23:02.03 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 23:03.19 | *** join/#asterisk De_Mon (de_mon@fl-67-232-63-136.dhcp.embarqhsd.net) |
| 23:13.10 | stephenw | Any ideas? |
| 23:14.18 | *** join/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 23:14.36 | z4nD4R | hi some tutorial how to implement zrtp on asterisk? |
| 23:15.10 | russellb | zRTP is not supported by Asterisk. |
| 23:17.01 | quintana | z4nD4R, http://zfoneproject.com/docs/asterisk/man/html/u_guide.html |
| 23:17.02 | z4nD4R | russellb: http://www.zfoneproject.com/prod_asterisk.html ? |
| 23:17.35 | z4nD4R | quintana: this look better, not? http://www.zfoneproject.com/prod_asterisk.html |
| 23:18.05 | russellb | not officially supported |
| 23:18.47 | quintana | and the patch is for asterisk 1.4 |
| 23:18.49 | pabelanger | teehee; 1-888-DIGIUM1 |
| 23:19.15 | z4nD4R | quintana: does not work with 1.8? |
| 23:20.13 | Kobaz | the girl on the main page is pretty hot |
| 23:20.14 | quintana | no i don't think but maybe you can send an email to the author ? |
| 23:20.46 | z4nD4R | quintana: u right i try it |
| 23:26.56 | z4nD4R | i find... that zrtp work not so gut like srtp.. is this true? |
| 23:28.09 | stephenw | I've got a very simple asterisk configuration. |
| 23:28.14 | stephenw | I'm trying to get a conference bridge working, using ConfBridge |
| 23:28.19 | stephenw | But asterisk isn't sending out any RTP. |
| 23:28.25 | stephenw | I've verified with tcpdump that it is receiving RTP from the UA |
| 23:28.34 | stephenw | But it isn't sending out any RTP packets. |
| 23:28.37 | stephenw | Tonnes of: Got RTP packet from 64.247.129.37:49860 (type 18, seq 023260, ts 3076334795, len 000020) |
| 23:28.43 | stephenw | See that plenty of times with rtp debug on |
| 23:28.47 | stephenw | but never any outgoing |
| 23:30.06 | *** join/#asterisk provolone (~root@c-68-33-201-70.hsd1.md.comcast.net) |
| 23:35.58 | *** part/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 23:50.48 | stephenw | I've got a very simple asterisk configuration. |
| 23:50.50 | stephenw | I'm trying to get a conference bridge working, using ConfBridge |
| 23:50.51 | stephenw | But asterisk isn't sending out any RTP. |
| 23:50.53 | stephenw | I've verified with tcpdump that it is receiving RTP from the UA |
| 23:50.54 | stephenw | But it isn't sending out any RTP packets. |
| 23:50.57 | stephenw | Tonnes of: Got RTP packet from 64.247.129.37:49860 (type 18, seq 023260, ts 3076334795, len 000020) |
| 23:51.00 | stephenw | See that plenty of times with rtp debug on |
| 23:51.03 | stephenw | but never any outgoing |
| 23:51.08 | stephenw | any ideas? |