| 00:00.33 | ChannelZ | I guess double check your custom directory permissions are readable/accessable by asterisk and that your file isn't bogus |
| 00:01.00 | mazpe | where are the default sound files.. like the ones for Authenticate? |
| 00:01.18 | mazpe | i'll copy one of those.. and test it in the custom folder |
| 00:01.19 | ChannelZ | core show settings |
| 00:01.38 | ChannelZ | it's either VarLib or Data but I don't remember (they might be the same) |
| 00:03.35 | mazpe | so what sets playback/background sounds to /var/lib/asterisk/sounds? |
| 00:04.41 | ChannelZ | the (pretty sure) VarLib directory shown on settings (via asterisk.conf) plus 'sounds' plus the language prefix if enabled |
| 00:06.55 | *** join/#asterisk sourcode (~code@ppp-58-8-205-221.revip2.asianet.co.th) |
| 00:13.26 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
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| 00:23.24 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 00:28.27 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 00:39.22 | mazpe | if i put the files in the /var/lib/asterisk/sounds it works.. if i make a directory like gbpbx or something it doesnt. |
| 00:39.31 | mazpe | directories need special permission or something? |
| 00:55.57 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 00:55.57 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 00:58.18 | *** join/#asterisk Woody2143 (~Woody2143@machine76.Level3.com) |
| 01:13.43 | gruvfunk | mazpe: if you create directories, check their permissions to match sounds dir perms |
| 01:16.10 | gruvfunk | best voip desk phone for a new office implementation with 15 people, any recommendations? |
| 01:19.05 | atan | gruvfunk, I quite like Cisco phones but Polycom are also popular. |
| 01:20.01 | gruvfunk | atan: is configuration simple? or does one need a TFTP server and such? |
| 01:21.08 | WIMPy | With 15 phones you might want one |
| 01:21.40 | gruvfunk | WIMPy: good point |
| 01:32.19 | m_tadeu | hi, I keep getting this when executing from an AGI..."handle_exec: Could not find application (Set(CDR(accountcode)=m_tadeu))"...isn't this the way I should set the account code? |
| 01:35.39 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 01:36.58 | gruvfunk | m_tadeu: try removing the first set of parenthesis "Set(CDR(accountcode)=m-tadeu)" |
| 01:39.24 | m_tadeu | gruvfunk: those parethesis are not mine...they are set my taken from the asterisk console....what I'm senging is ""EXEC Set(CDR(accountcode)=m_tadeu) |
| 01:43.36 | gruvfunk | m_tadeu: so is EXEC the right AGI command you want to use? |
| 01:44.24 | m_tadeu | gruvfunk: starting to doubt :) how should I set the accountcode from an agi script? |
| 01:45.34 | russellb | it's there a set variable or something? |
| 01:45.41 | russellb | "SET VARIABLE CDR(accountcode) foo" |
| 01:45.46 | gruvfunk | m_tadeu: well, my AGI-fu is poo, but perhaps you want "SET VARIABLE ... |
| 01:46.07 | gruvfunk | russellb: :) |
| 01:46.19 | russellb | EXEC should work if you get the syntax right |
| 01:46.27 | m_tadeu | gonna try that....thanx guys |
| 01:46.38 | *** join/#asterisk g00gle (~thameema@c-98-248-232-219.hsd1.ca.comcast.net) |
| 01:46.39 | russellb | it might be ... "EXEC Set CDR(accountcode)=foo |
| 01:46.59 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 01:47.00 | kaldemar | agi show commands |
| 01:47.15 | kaldemar | set variable it is |
| 01:50.45 | m_tadeu | set variable worked in deed :) thanx again |
| 01:59.27 | russellb | yay |
| 01:59.54 | jaytee | yippee! |
| 02:03.41 | Kobaz | oh wow |
| 02:03.46 | Kobaz | i never even knew about agi show commands |
| 02:03.55 | russellb | heh |
| 02:04.05 | *** join/#asterisk TJNII (~TJNII@207.189.199.62) |
| 02:09.49 | *** join/#asterisk jpsharp (~jsharp@74-95-145-86-Naples.hfc.comcastbusiness.net) |
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| 02:24.02 | *** join/#asterisk shmaltz (~chatzilla@mail.dmaven.com) |
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| 02:24.12 | shmaltz | is happe samba config is over |
| 02:24.59 | shmaltz | anyone here ever had to go thru the painful process of setting up samba on slackware? |
| 02:25.33 | shmaltz | hi everyone, anyone awake? |
| 02:26.14 | shmaltz | sits down in the side with his beer |
| 02:26.22 | shmaltz | is all lonly |
| 02:26.56 | Kobaz | come code with us |
| 02:26.59 | Kobaz | and you wont be lonly |
| 02:27.08 | Kobaz | lonely |
| 02:28.09 | shmaltz | Kobaz who/what is us? |
| 02:28.16 | shmaltz | as in the USA? :P |
| 02:28.40 | Kobaz | us as in, asterisk people |
| 02:28.45 | Kobaz | from all over |
| 02:30.05 | shmaltz | this is interesting: |
| 02:30.07 | shmaltz | http://www.latimes.com/news/local/sc-dc-0524-court-prisons-web-20110523,0,2337401.story |
| 02:33.54 | *** join/#asterisk dozment (~dan@173-165-163-145-atlanta.hfc.comcastbusiness.net) |
| 02:34.50 | ChannelZ | Yeah because CA wasn't f*ed up enough. |
| 02:35.03 | dozment | slaps doulos1 around a bit with a large trout |
| 02:36.01 | shmaltz | now if these inmates will move out of state they will get arrested again waiting extradition for ever |
| 02:36.07 | shmaltz | so I should be safe here in NJ |
| 02:37.27 | shmaltz | i guess this is a new definition in: money is blood: |
| 02:37.29 | shmaltz | http://hosted2.ap.org/COGRA/APWorldNews/Article_2011-05-23-EU-Bulgaria-Blood-for-Sale/id-c6cbbf2d7aa4400286441477b3a38ffa |
| 02:37.44 | Kobaz | heh |
| 02:39.01 | shmaltz | polycom rocks |
| 02:41.08 | *** part/#asterisk dozment (~dan@173-165-163-145-atlanta.hfc.comcastbusiness.net) |
| 02:41.57 | Kobaz | anyone use shared lines with cisco spa phones |
| 02:42.46 | shmaltz | kobaz, why shared lines? |
| 02:43.02 | Kobaz | because that's what people want these days |
| 02:43.26 | shmaltz | nah, buddy watching and follow me should do the trick |
| 02:43.32 | Kobaz | nope |
| 02:43.36 | shmaltz | why not? |
| 02:43.40 | Kobaz | because that's what people want |
| 02:43.41 | Kobaz | heh |
| 02:43.50 | shmaltz | ok, any better answer? |
| 02:45.04 | shmaltz | what functionality do you gain by shared over what I mentioned? |
| 02:45.06 | Kobaz | you have say, a boss/secretary setup |
| 02:45.16 | shmaltz | right and what I said accomplishes that |
| 02:45.23 | Kobaz | and you wont want call parking or transfering |
| 02:46.02 | shmaltz | i got you |
| 02:48.48 | Kobaz | just hit a button and the call is on your phone |
| 02:49.00 | Kobaz | people are used to 30 year old phone systems |
| 02:49.08 | Kobaz | and have been doing the same thing for years and dont want to learn something new |
| 02:49.28 | Kobaz | i have it kidn of faked with some polycom blf and some custom stuff |
| 02:49.38 | Kobaz | but true shared lines would be good.. i think the cisco supports it |
| 02:54.03 | jpsharp | DTMF question. If I have a provider with broken DTMF decoding talking to Asterisk talking to an ATA-186, Asterisk should still be able to do DTMF detection from the ATA side, right? |
| 02:56.01 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 02:56.34 | WiretapWork | jpsharp, that is spectacularly vague |
| 02:56.45 | WiretapWork | Asterisk only supports RFC2833 DTMF properly |
| 02:57.12 | Kobaz | spectacularly! |
| 02:58.25 | jpsharp | Inbound from provider to Asterisk to ATA, no DTMF detection anywhere, no matter inband, rfc2833, or SIP Info. |
| 02:58.46 | jpsharp | outbound from ATA to Asterisk to a different provider, DTMF works splendidly using rfc2833. |
| 02:59.49 | jpsharp | Makes using ## to transfer inbound calls difficult. |
| 03:00.37 | jpsharp | Even if the inbound provider has broken DTMF detection, Asterisk still should be able to detect DTMF from the ATAs? |
| 03:01.12 | Kobaz | not necessarily |
| 03:01.24 | Kobaz | dtmf in audio over voip protocols can be good or not good |
| 03:01.53 | shmaltz | jpsharp what codec? |
| 03:02.00 | jpsharp | ulaw. |
| 03:02.36 | jpsharp | I'd really like to run rfc2833. Inband audio sucks. |
| 03:04.28 | jpsharp | I don't understand why it isn't using 2833 while talking to the ATAs, even if it is using something else to talk to the provider. |
| 03:08.45 | jpsharp | ahha. Had to twiddle reinvite settings to keep Asterisk in the media stream. |
| 03:08.55 | jpsharp | It was never getting the RTP streams. |
| 03:09.40 | jpsharp | Sometimes you just have to explain the problem to someone to figure it out. |
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| 03:13.11 | saliak | My system seems to be the subject of an attack. Well, it was the subject of a successful attack and once I realized it I changed the pw on the compromised extension. Now that external system is still poking at extension. I'd like to use fail2ban to fix this. Is there a way to add the ip address that's failing authentication on the log message "Failed to authenticate device ....." from chan_sip.c? |
| 03:14.24 | WIMPy | That's what it's there for. |
| 03:14.35 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 03:14.54 | ChannelZ | fail2ban can be made to just read your asterisk log, no need to hack the channel driver |
| 03:19.21 | saliak | well my log file doesn't indicate the IP address |
| 03:20.06 | saliak | "Failed to authenticate device "17656107745" <sip:100@68.9.228.184>;tag=as21e3976d" |
| 03:20.08 | saliak | oh |
| 03:20.12 | saliak | crap, sorry, it does |
| 03:20.17 | saliak | the ip is very close to my actual one |
| 03:20.23 | saliak | off by a single digit, so I just misread it |
| 03:20.41 | saliak | my bad |
| 03:21.15 | ChannelZ | damn cable users |
| 03:22.47 | *** join/#asterisk Deeewayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 03:22.47 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 03:27.16 | saliak | yeah, i'm so naive. i had this silly notion that no one would care about a random computer on the internet. so someone used my system to dial numbers incrementally. i guess there's no way to know what content was going across the call? |
| 03:28.09 | WiretapWork | saliak, only if you had call recording on, which I have as a matter of course :P |
| 03:29.05 | saliak | haha, that's a little creepy for me, but yeah, i guess that leaves us in a lurch in situations like these. it is usually telemarketing? |
| 03:29.39 | WIMPy | international calling card business |
| 03:31.49 | jpsharp | Or hackers calling their friends in Ukraine or Romania. |
| 03:31.56 | jpsharp | yeah, what he said. |
| 03:32.02 | shmaltz | I was listened in on one of these, it was one of the car warranty scams |
| 03:32.21 | shmaltz | i used chanspy |
| 03:32.52 | saliak | ah |
| 03:33.02 | saliak | I've gotten the car warranty calls before |
| 03:33.12 | jpsharp | I had someone root my * box, go sniffing through my sip.conf, grab the config for an ATA account, and then use it to make $150 worth of calls to the UK and Romania. |
| 03:33.52 | shmaltz | root * asterisk is a way more sophisticated attack than sip brute forcing, how did they root it? |
| 03:35.06 | ChannelZ | 1. If you're going to leave your box sitting in the open, consider not having numeric SIP device names; 2. generate good passwords for said devices; 3. really think about what you allow to happen in your dialplan. context is everything |
| 03:35.37 | ChannelZ | (this of course not accounting for someone breaking into the box outright and looking at your config in which case nothing matters at that point.) |
| 03:35.49 | shmaltz | also don't forget there is no reason to: 1. setup sshd on default port. 2. allow from any host. 3. allow root access. |
| 03:35.57 | *** join/#asterisk sabgenton (~chatzilla@238.162.dsl.mel.iprimus.net.au) |
| 03:36.06 | jpsharp | Oddly enough, I got hacked through a hole in the festering pile of crap known as Wordpress. |
| 03:36.21 | WiretapWork | shmaltz, indeed, the only box on my network which has SSH accessible doesn't have it on the default port |
| 03:36.24 | shmaltz | why would you have it completely open? |
| 03:36.35 | WiretapWork | also, mixing webserver and asterisk is a bad mix anyway :P |
| 03:36.35 | ChannelZ | and turn off SSH passwords totally unless you can't bring a thumbdrive with an SSH key with you |
| 03:36.37 | shmaltz | wiretapwork, I always change the port number |
| 03:36.41 | jpsharp | Its what I get for not separating services onto different machines. |
| 03:37.04 | WiretapWork | jpsharp, should also make sure that internal-only extns can't dial from outside |
| 03:37.10 | shmaltz | ChannelZ, I never did it so far, reason being that so far I only saw script kiddies doing the work which relies on the default port |
| 03:37.20 | WiretapWork | shmaltz, I meant to say no root access, no idea why I said non-default port, as I _do_ use Port22 |
| 03:37.26 | sabgenton | does anyone know if sofia sip gives 729 suport ? |
| 03:37.37 | sabgenton | <PROTECTED> |
| 03:37.41 | shmaltz | sabenton, what is sofia sip? |
| 03:37.43 | WiretapWork | sabgenton, this is #asterisk not #freepbx |
| 03:37.44 | WiretapWork | err |
| 03:37.47 | sabgenton | shows g729 as an option |
| 03:37.47 | WiretapWork | #freeswitch |
| 03:37.51 | saliak | jpsharp - yeah, i'm in that situation now. i have everything on one machine. it appeals to the conservationist in me |
| 03:37.54 | ChannelZ | yeah. I used to have denyhosts running (well, still do actually) and it would block dozens of IPs a day for SSH hacking. Moved SSH off 22 and I haven't gotten a single attempt since |
| 03:37.57 | jpsharp | I used the whole clusterflop to rebuild my network. I drunk the VMWare koolaid deeply. A dozen VMs now instead of a monolithic system. |
| 03:38.19 | WiretapWork | jpsharp, I have 8 VMs and 3 physicals :P |
| 03:38.24 | sabgenton | WiretapWork: yay thaks was looking allover for a chanel |
| 03:38.39 | WiretapWork | sabgenton, freeswitch in an asterisk channel... really? |
| 03:38.59 | jpsharp | blasphemy! |
| 03:39.12 | shmaltz | WiretapWork, the other day I someone at #linux asking how to get rid of a BSOD |
| 03:39.17 | ChannelZ | I only said 'this halibut is good enough for Jehova' |
| 03:39.38 | WiretapWork | shmaltz, LOL |
| 03:39.39 | sabgenton | WiretapWork: oh then why did you mention it |
| 03:39.41 | jpsharp | shmaltz: Told em to get a Mac? |
| 03:40.06 | shmaltz | I don't like Mac, its too expensive |
| 03:40.06 | ChannelZ | sounds like a self-answering question given the venue |
| 03:40.08 | WiretapWork | sabgenton, this is #asterisk, freeswitch is the PBX that calls its sip module sofia...... and you're not going to find that here, you'll find it in #freeswitch |
| 03:40.11 | shmaltz | very nicely designed though |
| 03:40.15 | sabgenton | I just don't know where to find informationo about linux voip codecing |
| 03:40.44 | sabgenton | WiretapWork: ah |
| 03:40.58 | shmaltz | sabgenton, try #2600 |
| 03:41.00 | sabgenton | sofia was inventied by nokia I understood |
| 03:41.16 | ChannelZ | which also has nothing to do with Asterisk |
| 03:41.17 | sabgenton | <PROTECTED> |
| 03:41.47 | sabgenton | people with sofia might be able to help me |
| 03:41.52 | shmaltz | is dancing around circles with a beer in each hand laughing his a** off |
| 03:43.14 | sabgenton | ooeky |
| 03:43.31 | shmaltz | somone help me out, is this really necessary? |
| 03:43.33 | shmaltz | http://en.wikipedia.org/w/index.php?title=Bit&action=historysubmit&diff=430619249&oldid=430109467 |
| 03:44.28 | jpsharp | That's not even the example of a bit. That's more than a bit. |
| 03:44.46 | shmaltz | jsharp good point, reverting it now |
| 03:44.56 | WiretapWork | lol |
| 03:45.13 | WiretapWork | that's spectacularly silly change |
| 03:45.20 | shmaltz | ok, same guy just reverted it: |
| 03:45.22 | shmaltz | http://en.wikipedia.org/w/index.php?title=Bit&diff=next&oldid=430619249 |
| 03:46.06 | shmaltz | ok, my battery of my BB died, its time to go home |
| 03:46.08 | shmaltz | cya guys |
| 03:46.11 | shmaltz | gnite |
| 03:46.44 | shmaltz | sabgenton, sorry sending you to #2600 wasnt meant to make fun of you, I figured you would know not to go there |
| 03:57.26 | sabgenton | oa |
| 03:58.49 | sabgenton | so what in the geek does it mean? |
| 03:59.09 | sabgenton | <PROTECTED> |
| 03:59.36 | jpsharp | 2600 is the name of the group where hackers & phreakers hang out. |
| 04:09.42 | atan | :D |
| 04:11.31 | sabgenton | heh |
| 04:11.43 | sabgenton | wow all four of them |
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| 04:50.03 | saliak | anyone use fail2ban with shorewall? for some reason, the "shorewall drop" command that fail2ban calls when i get my requisite # of failed connections doesn't seem to actually stop them |
| 04:54.35 | ChannelZ | I don't use shorewall - does that command add a rule to iptables live, or just to a config file for the next time it's reloaded? |
| 04:55.20 | ChannelZ | also there could be a consideration of the user fail2ban runs the command as and whether it can in turn modify the firewall at all |
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| 07:22.26 | cneb3000 | good morning, UK |
| 07:22.47 | wdoekes2 | only UK? |
| 07:22.55 | cneb3000 | yea, problem? ;) |
| 07:22.58 | tuxx- | thats not fair, i want a good morning too :-( |
| 07:23.06 | wdoekes2 | goeiemorgen NL! |
| 07:23.09 | robbie` | its only 12:30AM in california, still a good night :P |
| 07:23.10 | tuxx- | idd :-) |
| 07:23.11 | cneb3000 | "good morning world" sounded a little cheesy! |
| 07:23.21 | tuxx- | Tue May 24 09:23:21 CEST 2011 |
| 07:23.26 | tuxx- | :-P |
| 07:23.29 | cneb3000 | hehe |
| 07:26.51 | Maliuta | not morning here, I'm at GMT+10 |
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| 07:33.06 | cneb3000 | to those where it's 23:00:00 or later.. go to bed! ;) |
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| 07:37.35 | ChannelZ | You're not the boss of me! |
| 07:37.46 | cneb3000 | :D |
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| 07:56.32 | schmidts | good morning |
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| 08:11.18 | voiptelecom | hello world ! |
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| 08:29.39 | skrusty | morning |
| 08:35.28 | cneb3000 | howdy do |
| 08:45.49 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:c033:6253:209c:7855) |
| 08:51.52 | *** join/#asterisk Tim_Toady (~moi@188.4.51.59) |
| 09:08.45 | *** join/#asterisk MarKsaitis (~MarKsaiti@host81-137-62-5.in-addr.btopenworld.com) |
| 09:13.11 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 09:32.00 | voiptelecom | I used "iax show peers" to know IP adress connect on my iax account |
| 09:32.19 | *** join/#asterisk Cadey (~x@62.84.178.106) |
| 09:32.45 | voiptelecom | I've got some account with many IP adresse connected on my IAX account |
| 09:33.09 | voiptelecom | how can i have the list of the IP adresse connected on my iax account ? |
| 09:33.21 | Cadey | Hi guys, stupid question but does setting the CallerID(num) and (name) for an outgoing call send the withheld signal to an outgoing call is anonymums |
| 09:33.57 | Cadey | *Anonymous |
| 09:34.06 | voiptelecom | cadey do you want anonymous calls ? |
| 09:34.19 | Cadey | we want to be able to make some yes |
| 09:34.34 | cneb3000 | Morning Cadey ;) |
| 09:34.43 | Cadey | Morning cneb :) |
| 09:34.54 | Cadey | you didnt add me to MSN did you? |
| 09:35.08 | cneb3000 | ....been a while since I used MSN and cant remember my password '¬_¬ |
| 09:35.13 | Cadey | hah |
| 09:35.18 | Cadey | what you use these days |
| 09:35.29 | cneb3000 | skype mainly |
| 09:35.37 | Cadey | dirty skype |
| 09:35.38 | Cadey | :P |
| 09:35.44 | cneb3000 | hey! it has its moments ;) |
| 09:35.50 | Cadey | lol |
| 09:35.51 | voiptelecom | CallerId(Anonymous) don't work ? |
| 09:35.53 | kaldemar | Cadey: "core show function CALLERID" will list you *-pres datatypes. you can set those. for values, see "core show function CALLERPRES". |
| 09:36.20 | Cadey | CallerId(Anonymous) <-- have not tried that |
| 09:36.26 | Cadey | looks |
| 09:36.28 | kaldemar | it will not work. |
| 09:36.36 | Cadey | oh ok Lakdemar |
| 09:36.46 | Cadey | ill take a look at your words |
| 09:38.00 | voiptelecom | i think it does |
| 09:38.25 | voiptelecom | i had tried it and got good result |
| 09:38.56 | Cadey | Anonymous isnt listed as a datatype voiptelecom ? |
| 09:39.48 | Cadey | kaldemar, so it looks like you do this... CallerId(name-pres) = unavailable |
| 09:39.59 | Cadey | CallerId(num-pres) = unavailable |
| 09:40.00 | Cadey | too |
| 09:40.44 | kaldemar | or one of the prohibited ones. try it out. |
| 09:41.08 | Cadey | I will take a look :) |
| 09:42.38 | kaldemar | remember that function names are case sensitive. so it's CALLERID, not CallerId. |
| 09:44.46 | voiptelecom | CALLERID(name)=Anonymous |
| 09:46.11 | voiptelecom | do you have any idea for my problem ? |
| 09:49.37 | *** join/#asterisk ihor (~Miranda@194.44.15.90) |
| 09:49.39 | *** join/#asterisk ironm (~ironm@fwj00.e-fon.ch) |
| 09:49.47 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:50.35 | kaldemar | voiptelecom: i don't understand what your problem is. |
| 10:00.56 | voiptelecom | I've got some IAX account with many IP adresse connected |
| 10:01.05 | voiptelecom | how can i have the list of the IP adresse connected on my iax account ? |
| 10:01.19 | voiptelecom | "iax show peers" show only 1 adress |
| 10:02.08 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 10:02.47 | Azrael808 | Hi, does anybody here have a good recommendation for a wifi SIP phone? |
| 10:02.54 | kaldemar | voiptelecom: you can't have many ip addresses, just one. |
| 10:03.08 | voiptelecom | i have many ip adresse |
| 10:03.18 | cneb3000 | Azrael: Wifi SIP phone? Like a physical one or one that runs on a computer/phone?.. |
| 10:03.27 | voiptelecom | i would have the list of it |
| 10:03.37 | Azrael808 | Physical one |
| 10:03.49 | voiptelecom | some clients use many serveurs for 1 IAX account |
| 10:03.56 | *** join/#asterisk coppice (~chatzilla@122.225.226.140) |
| 10:04.09 | cneb3000 | Azrael: So a phone that doesn't need wiring up.. just to be clear? |
| 10:04.11 | kaldemar | voiptelecom: you CAN'T. only the last registration source is shown. |
| 10:04.27 | voiptelecom | there is always a solution... |
| 10:04.29 | voiptelecom | =) |
| 10:05.00 | kaldemar | voiptelecom: asterisk does not support multiple registrations. you need to make more than one definition in iax.conf and make the clients use different accounts. |
| 10:05.08 | Azrael808 | cneb3000: yup, I was thinking of a device that spoke 802.11b/g to connect to the office wifi. |
| 10:05.13 | voiptelecom | for cdr it work |
| 10:05.27 | Azrael808 | So essentially, if there were any cables, they would simply be to charge the device |
| 10:05.42 | voiptelecom | i've make a config and i know, when i have more than 1 ip adress login on 1 IAX account |
| 10:06.05 | voiptelecom | cdr told me witch adress make the call |
| 10:06.26 | voiptelecom | but i would like to know how have this information in real time |
| 10:06.38 | voiptelecom | kaldemar |
| 10:06.39 | kaldemar | voiptelecom: asterisk still holds information for one single address only at the same time. |
| 10:06.45 | cneb3000 | Azrael: I personally wouldn't rely on Wifi unless the connection was /n myself. You dont tend to get VoIP on wifi simply because it just isnt as reliable as ethernet cables :) |
| 10:06.47 | voiptelecom | it's not support but it's worked ? |
| 10:07.20 | Azrael808 | Well, we do have a /n network available... Just figured the phones would be that much more expensive! |
| 10:07.21 | Azrael808 | :) |
| 10:07.25 | voiptelecom | actually i have some client who used many adress ip for the same account |
| 10:07.34 | cneb3000 | i bet any wifi phone is expensive to be honest :P |
| 10:07.38 | Azrael808 | Still, plz give me your recommendation :) |
| 10:07.41 | voiptelecom | iax show peer show only 1 adresse |
| 10:07.52 | Azrael808 | Yeah, I've seen 'em for around GBP 170 |
| 10:07.57 | Chainsaw | Azrael808: I would recommend a DECT phone with the base speaking SIP, on ethernet. |
| 10:08.09 | robbie` | anyone familiar with avantfax? i can send/recv faxes but the incoming faxes don't appear in the inbox, i can find them in the hylafax recvq though |
| 10:08.13 | voiptelecom | but in my cdr i know witch adress make a call |
| 10:08.15 | Chainsaw | Azrael808: Siemens makes them. |
| 10:08.18 | kaldemar | voiptelecom: sure calls can be made from different addresses if the authentication credentials are correct. but still, asterisk will only keep one address for a registered peer. |
| 10:08.36 | voiptelecom | ok |
| 10:09.03 | voiptelecom | is it possible to change it ? |
| 10:09.22 | kaldemar | voiptelecom: registration is just a way to let the other end know where you are. it's not a login type of thing. |
| 10:09.29 | voiptelecom | is there is a risk with many authentication on 1 account ? |
| 10:09.34 | Azrael808 | Chainsaw: I don't know much about DECT, is there any chance of interference with wifi? |
| 10:09.43 | Chainsaw | Azrael808: None. Different frequency spectrum. |
| 10:09.56 | Azrael808 | OK cool |
| 10:09.57 | voiptelecom | bevause some client ask 10 IP adresse WAN |
| 10:09.57 | kaldemar | voiptelecom: you can change it by changing chan_iax2.c (and others). |
| 10:10.01 | voiptelecom | i dont want to make 10 account |
| 10:10.13 | Azrael808 | OOI, are there any models you'd recommend? |
| 10:10.18 | kaldemar | voiptelecom: no risk, but only 1 of them can receive calls. make 10 accounts. |
| 10:10.26 | Azrael808 | That's to both Chainsaw and cneb3000 :) |
| 10:10.27 | voiptelecom | ok |
| 10:10.37 | Chainsaw | Azrael808: I have a C450 in use. |
| 10:10.39 | voiptelecom | we have only outgoing call on serveurs |
| 10:11.04 | Chainsaw | Azrael808: The others are an earlier type. It's mostly about whether you want an analog line as well, and whether you want an answering machine built in. |
| 10:11.09 | voiptelecom | what can you recommand me ? |
| 10:11.20 | voiptelecom | many account or dev for solve my pb ? |
| 10:11.49 | Azrael808 | Chainsaw: a basic SIP client is all I need, our Asterisk server does VM :) |
| 10:12.00 | kaldemar | voiptelecom: obviously many accounts, as i have twice already said. |
| 10:12.45 | voiptelecom | ok thx |
| 10:13.25 | voiptelecom | really not functionnal for 1 client who ask me 1 account and 10 @IP |
| 10:13.47 | voiptelecom | i don't understand no easier solution for asterisk |
| 10:14.02 | voiptelecom | well thx a lot |
| 10:14.06 | voiptelecom | g2g eat !! |
| 10:14.08 | voiptelecom | su |
| 10:14.17 | voiptelecom | afternoon |
| 10:14.59 | cneb3000 | Azrael: Sorry I couldn't be more help! |
| 10:15.17 | Azrael808 | cneb3000: no probs, I've got some good info to go on now thx! |
| 10:15.44 | Azrael808 | I'll probably do some reliability testing with a SIP client on my Android phone anyways and put some options out for the powers that be ;) |
| 10:16.13 | cneb3000 | reliability testing? surely they'll go for whatever looks best? ;) |
| 10:16.26 | *** join/#asterisk irroot (~gregory@dsl-185-122-118.dynamic.wa.co.za) |
| 10:16.51 | Azrael808 | LOL... True, true, wouldn't be surprised if they just decided to use their iPhone 4s! |
| 10:17.09 | coppice | the SIP client built into android werks very well, but doesn't support any wideband codecs |
| 10:21.20 | irroot | have a customer wanting to go 100% wifi converged will likely get all staff droid phones |
| 10:30.09 | irroot | ok getting a few cases where sip_write is bouncing frames with incompat formats this never used to happen i added some magic to set the write format in a attempt to get the audio up this good / bad /ugly ?? |
| 10:30.23 | irroot | see r320709 |
| 10:36.29 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 10:39.21 | *** join/#asterisk GreatSUN (~greatsun@188-22-190-29.adsl.highway.telekom.at) |
| 10:39.23 | GreatSUN | re |
| 10:40.35 | GreatSUN | does someone have an idea, why one of my two asterisk servers can contact the other and the other says "unreachable" |
| 10:40.38 | GreatSUN | ? |
| 10:40.40 | GreatSUN | OfficeFe/6000 188.20.12.162 N 5060 OK (58 ms) |
| 10:40.53 | GreatSUN | OfficeHi/6001 188.22.190.29 N 5060 UNREACHABLE |
| 10:41.14 | GreatSUN | nat=yes, canreinvite=yes,qualify=yes |
| 10:41.23 | GreatSUN | port is standard udp 5060 |
| 10:41.45 | GreatSUN | firewall is set up to allow all connections |
| 10:42.15 | wdoekes2 | GreatSUN: use tcpdump on the 188.22.190.29 machine to see if any packets are received at all |
| 10:42.38 | wdoekes2 | and continue debugging the problem from there |
| 10:44.43 | GreatSUN | 12:44:20.444996 IP pbx1.intern.scv.co.at.sip > 188.20.12.162.sip: SIP, length: 520 |
| 10:44.44 | GreatSUN | 12:44:20.592819 IP 188.20.12.162.sip > pbx1.intern.scv.co.at.sip: SIP, length: 493 |
| 10:44.48 | GreatSUN | thats all |
| 10:45.09 | Azrael808 | coppice: which SIP client do you mean? |
| 10:45.45 | coppice | Azrael808: the one built into recent versions of android |
| 10:46.10 | GreatSUN | Corydon76-home: there is a siip-client in 2.3.4? |
| 10:46.13 | *** join/#asterisk Faithful (~Faithful@1.152.98.127) |
| 10:46.29 | Azrael808 | Ah, ok the phone I'm testing on is 2.1 and I'm using SIPDroid (seems pretty usable atm) |
| 10:46.31 | GreatSUN | Corydon76-home: sorry |
| 10:46.38 | Azrael808 | What version of Android has built in client? |
| 10:46.44 | GreatSUN | coppice: there is a sipclient in 2.3.4? |
| 10:47.02 | GreatSUN | Azrael808: in 2.3.3 there is none afaik |
| 10:47.07 | coppice | yes. its very well integrated with the dialer |
| 10:47.09 | GreatSUN | I have 2.3.3 running |
| 10:47.27 | irroot | the sip libs are in android from 2.3 |
| 10:47.31 | Azrael808 | Dammit, just sent off my Desire HD for repair, so I can't check lol |
| 10:47.36 | irroot | the client is added after that |
| 10:47.36 | GreatSUN | coppice: you got a nexus S dont ya? |
| 10:48.08 | GreatSUN | irroot: does it matter if the libs are there, but the client isnt? |
| 10:48.13 | coppice | yes. I wish I didn't, but I do |
| 10:48.24 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 10:48.35 | Azrael808 | coppice: why don't you like your Nexus S? |
| 10:48.37 | irroot | GreatSUN there is a demo sip client in the dev tools |
| 10:48.44 | GreatSUN | ah ok |
| 10:48.51 | GreatSUN | don't have dev-tools |
| 10:49.00 | GreatSUN | but wished I had 2.3.4 |
| 10:49.19 | coppice | I thought it would be safe to buy a samsung phone when they were not in control of the software. mistake |
| 10:49.41 | GreatSUN | but anyways |
| 10:49.53 | GreatSUN | does anyone have an idea what I could do about my problem? |
| 10:50.25 | GreatSUN | I bet there is some configuration issue in the 188.20.12.162 system |
| 10:50.33 | GreatSUN | but I don't know where to search |
| 10:54.12 | irroot | GreatSun what errors happen if you sip debug on the box |
| 10:54.23 | irroot | when you sip reload ?? |
| 10:54.27 | GreatSUN | irroot: on the 188.20.12.162 box? |
| 10:54.48 | irroot | the one you pasted above |
| 10:54.54 | irroot | that cant reach other |
| 10:55.01 | puzzled | tzafrir: do you have a dahdi-zaphfc patch in your debian tree I can look at so I can figure out how to make it compile zahfc? |
| 10:55.30 | GreatSUN | the one named pbx1... can reach the 188.20.12.162 |
| 10:55.49 | GreatSUN | I can also place calls there (didn't try audio, but ringing works) |
| 10:55.50 | irroot | also try this if using linux options nf_conntrack_sip sip_timeout=300 |
| 10:55.58 | irroot | in modprobe.conf |
| 10:56.09 | irroot | and run "conntrack -F" to flush it |
| 10:56.44 | irroot | that is if you load the sip nf conntrack helpers worth it |
| 10:57.50 | tzafrir | puzzled, it's generally part of dahdi-extra |
| 10:58.31 | tzafrir | http://gitorious.org/dahdi-extra |
| 10:58.32 | puzzled | tzafrir: excellent. is there a git repo I can have a look? |
| 10:58.52 | puzzled | heh, slow typing. thanks! |
| 11:10.48 | *** join/#asterisk ajkaanbal (~ajkaanbal@189.181.105.67) |
| 11:33.10 | *** join/#asterisk \DSAFEW\ (~\DSAFEW\@ip68-3-53-165.ph.ph.cox.net) |
| 11:39.18 | *** join/#asterisk marlowe (~marlowe@static-72-66-8-138.washdc.fios.verizon.net) |
| 11:41.41 | *** join/#asterisk marlowe (~marlowe@static-72-66-8-138.washdc.fios.verizon.net) |
| 11:46.08 | *** join/#asterisk wonderworld (~ww@port-92-201-90-244.dynamic.qsc.de) |
| 11:47.12 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 11:57.44 | tuxx- | hey guys, does the ami manager command 'Redirect' support multiple 'ExtraChannel' variables? |
| 11:57.52 | tuxx- | http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Redirect |
| 11:57.53 | tuxx- | :P |
| 12:00.44 | Cadey | Hi guys, do any of you have a script to split up the output of the logger into hour text files (or other periodic splitter) |
| 12:01.17 | mazpe | Any directions on how i can create a perl script that generates a call? |
| 12:01.30 | Cadey | So for example this could be split into many files instead of one file |
| 12:01.37 | Cadey | logger.conf -- /mnt/TAM01FPS02Calls/AsteriskCalls/sip.txt => verbose |
| 12:03.21 | mazpe | oh... i can $AGI->exe('Dia','SIP/XXXXXXXX'); |
| 12:03.55 | kaldemar | mazpe: make it call CLI originate with 'asterisk -rx ...', move a call file to spool dir or connect AMI and use the originate command. |
| 12:04.11 | cneb3000 | cadey: check this out - i use it somtimes ---> http://www.cyberciti.biz/faq/how-do-i-rotate-log-files/ |
| 12:04.31 | kaldemar | mazpe: nevermind, use a proper AGI way if it's an AGI script. |
| 12:04.33 | mazpe | kaldemar: a call file to the spool dir sounds interesting.. didnt even know there was such |
| 12:04.56 | mazpe | kaldemar: well its kind of a dialer that i'm building for a client. |
| 12:05.02 | *** join/#asterisk fish-bulb (~qcstewart@nat/digium/x-uvdjappmwrqtgovv) |
| 12:06.08 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 12:06.31 | cneb3000 | cadey: here's an asterisk specific guide ---> http://www.voip-info.org/tiki-index.php?page=logrotate |
| 12:07.55 | *** join/#asterisk cneb3000 (~ben.cropl@gateway.magneticnorth.com) |
| 12:08.09 | Cadey | cneb3000 : thats for the messages log dude |
| 12:08.22 | cneb3000 | 'salright! share the wealth! |
| 12:08.23 | Cadey | I want to output the verbose messages for a few days |
| 12:10.42 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 12:16.26 | tuxx- | is there any way to not let the ami command 'Park' announce which parkingslot it put the channel in? :P |
| 12:16.37 | tuxx- | without removing the soundfiles, hehe :P |
| 12:16.51 | leifmadsen | tuxx-: remove the...^H^H^H^H^H^H^H^H |
| 12:16.59 | tuxx- | ;-) |
| 12:17.18 | leifmadsen | I guess you could change the code? :) |
| 12:17.30 | mazpe | kaldemar: so you can only have 1 number per call file? |
| 12:17.33 | tuxx- | hmye, dont want to make changes to the asterisk source preferably, but if thats the only way :) |
| 12:17.59 | kaldemar | mazpe: no. |
| 12:19.28 | tuxx- | tnx leifmadsen :P |
| 12:19.50 | leifmadsen | I'm not aware of a "don't announce" option, but it sounds useful (when used from AMI) |
| 12:22.23 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 12:24.00 | Kobaz | tuxx-, leifmadsen: it's a hacky way, but in order to not announce the parking lot, set the dial channel for ParkAndAnnounce to Console/dsp |
| 12:24.09 | leifmadsen | oh neato |
| 12:24.16 | leifmadsen | hacky indeed, but probably effective |
| 12:24.20 | leifmadsen | tuxx-: ^^^ |
| 12:24.23 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 12:24.23 | tuxx- | ah nice :) |
| 12:24.25 | tuxx- | tnx =) |
| 12:33.35 | mazpe | anyone using flowroute? I'm looking for an inexpensive but good quality sip provider |
| 12:33.50 | leifmadsen | never heard of them |
| 12:33.59 | leifmadsen | feels deja vu |
| 12:34.21 | mazpe | they have a nice site and they seem to be around for a while. |
| 12:34.39 | leifmadsen | I like www.unlimitel.ca for quality and cheapness. I also know of people who use voip.ms successfully. |
| 12:34.57 | leifmadsen | I've also had good luck with bandwidth.com, but beyond that I can't recommend anyone else |
| 12:35.12 | mazpe | I currently use varphonex, but its like 0.025 |
| 12:35.23 | leifmadsen | unlimitel is 0.01 |
| 12:35.40 | mazpe | to the us? |
| 12:35.41 | leifmadsen | voip.ms is like 0.0167 in many areas, sometimes cheaper |
| 12:35.50 | leifmadsen | mazpe: to USA/Canada yes |
| 12:36.12 | leifmadsen | (except for some more expensive rate centres, but you can look that up on the rate table they provide) |
| 12:36.15 | mazpe | is the providers server location a deciding factor? |
| 12:36.22 | leifmadsen | it isn't for me |
| 12:36.39 | mazpe | I'm using linode atlanta center. |
| 12:36.53 | leifmadsen | latency from my server to their location could be a factor |
| 12:37.00 | voiptelecom | do you recommend a cheap sip provider for burkina faso ? |
| 12:37.41 | mazpe | leifmadsen: do you use a database or something to select your cheapest route? |
| 12:43.17 | leifmadsen | could |
| 12:43.20 | leifmadsen | don't necessarily |
| 12:47.03 | cneb3000 | Mazpe: the industry term for that is 'least cost routing' :) |
| 12:47.10 | mazpe | lcr |
| 12:47.11 | cneb3000 | Mazpe: or 'LCR' |
| 12:47.15 | cneb3000 | yes ;) |
| 12:47.28 | mazpe | I was using at a point a2billing to accomplish it. |
| 12:47.34 | mazpe | but it seem to much of a hassel |
| 12:48.18 | cneb3000 | you could probably include it in a dial plan.. not sure what that would do the load on the box. |
| 12:48.19 | cneb3000 | Every outbound call checks a database for the cheapest trunk to use.. |
| 12:48.39 | mazpe | and for some reason, forwarding calls didnt work at all between cisco 7960 phones and a2billing.. very odd. |
| 12:49.15 | cneb3000 | really :| |
| 12:49.37 | mazpe | sounds like a lot of load to differentiate between 0.0098 and 0.0015 |
| 12:49.46 | mazpe | for international calls i can certainly see it. |
| 12:49.59 | cneb3000 | I've never played with a2billing myself. I hear it mentioned a lot. Normally by entrepreneurs trying to set up 'calling card' boxes :) |
| 12:50.25 | cneb3000 | ^^^ yes true |
| 12:56.44 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:03.58 | *** join/#asterisk l2trace99 (~jr@74.118.40.1) |
| 13:04.30 | *** join/#asterisk MarKsaitis (~MarKsaiti@host81-137-245-117.in-addr.btopenworld.com) |
| 13:09.08 | irroot | a wise man once said beware of the _ at start of variable names |
| 13:09.41 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 13:10.23 | cneb3000 | ^^ hehe |
| 13:21.24 | Katty | hello my asterisk does not work at all how to fix pls |
| 13:21.47 | leifmadsen | echo "Ohai Russell. Please to be my fixing Asterisk." | mutt -s "Please to be fixing my stuff" russell@shifteight.org |
| 13:22.07 | *** join/#asterisk wesphillips (~wphill04@137.237.233.124) |
| 13:22.40 | irroot | Katty what you break .... |
| 13:22.45 | tuxx- | hehe |
| 13:22.53 | Katty | gooooood morning mister leif! |
| 13:22.56 | Katty | applies huggage to leifmadsen |
| 13:23.27 | leifmadsen | accepts huggage |
| 13:23.29 | leifmadsen | for a fee! |
| 13:23.36 | Katty | WHAT |
| 13:23.40 | b0ot | How difficult would it be to have lots of computers running asterisk talk to one another |
| 13:23.42 | leifmadsen | I charge you $0.02 |
| 13:23.44 | Katty | boy, we's gonna have words. |
| 13:24.03 | irroot | leif = love in afrikaans FYI |
| 13:24.06 | leifmadsen | b0ot: not particularly difficult, although "talk to each other" is ambiguous |
| 13:24.06 | cneb3000 | b0ot: like an asterisk botnet? |
| 13:24.12 | angryuser | b0ot, well it will depend on the topic of the chat |
| 13:24.49 | b0ot | well I mean if I have a very mobile network where people could me moving all over the place.... I'm trying to decide how many asterisk call managers I would want |
| 13:26.01 | Katty | so. |
| 13:26.07 | Katty | i'm going camping for the first time /ever/ |
| 13:26.11 | Katty | words of advice? |
| 13:26.21 | leifmadsen | Katty: bring sweaters for the campfire |
| 13:26.27 | leifmadsen | it'll be colder at night than you'd think |
| 13:26.40 | leifmadsen | also, everything you bring with you is going to smell of camp fire |
| 13:26.44 | Katty | i don't know that i own a sweater, but i'll bring a hoodie |
| 13:26.59 | angryuser | b0ot, mobile users does not mean you need many asterisk, and i have no idea why do you talk about call managers |
| 13:27.05 | b0ot | sunscreeen, knife, lighter, waterproof bag, |
| 13:27.24 | angryuser | ~topic |
| 13:27.42 | tuxx- | Katty: the only thing you need to survive is beer. |
| 13:27.48 | b0ot | angryuser, I mean that my network topology changes quite often |
| 13:27.56 | leifmadsen | Katty: bring layers |
| 13:28.07 | angryuser | b0ot, i still dont see any problem |
| 13:28.11 | leifmadsen | also a couple of towels and a couple swim suits if near water |
| 13:28.42 | b0ot | angryuser, in some circumstances the computer might not always have access to the computer running asterisk |
| 13:28.46 | b0ot | if i only had one running |
| 13:29.21 | angryuser | b0ot, why is that ? |
| 13:29.25 | Katty | Qwell: TAKE ME WITH YOU |
| 13:29.29 | irroot | katty last time i went camping it was in the zambezi valley no tents in the game reserve out in the wild no fences either :P |
| 13:29.41 | b0ot | I told you it is a mobile network... not everyone is in contact with everyone |
| 13:29.44 | b0ot | at the same time |
| 13:29.54 | Katty | irroot: something tells me i shouldn't be camping on a game reserve for the first time ever |
| 13:29.55 | angryuser | b0ot, have you heard about VPN ? |
| 13:30.08 | Katty | leifmadsen: the water looks icky. i don't wanna swim in it :< |
| 13:30.18 | b0ot | angryuser, not everyone would have access to internet |
| 13:30.23 | b0ot | just private network |
| 13:30.29 | Katty | leifmadsen: but i'm sure my doggy will get into it and have a lovely time. towel for him! |
| 13:31.08 | cneb3000 | b0ot: have the server display on a public AND internal IP?.. |
| 13:31.30 | angryuser | b0ot, in this case, why do you involve asterisk with the people without Network ? How in hell multiple asterisk help peoploe without net ? |
| 13:32.17 | *** join/#asterisk wonderworld (~ww@port-92-201-90-244.dynamic.qsc.de) |
| 13:32.27 | irroot | Katty nope not best idea going to sleep with sounds of hyena and lions in background pretty decent ... have a blast |
| 13:33.16 | b0ot | angryuser, so lets say I have three groups A, B, C. If A and B get cut off from C, at least they would still be able to talk to one another. Now if you increase that to more groups... and they somehow got seperated it would be nice if the respective groups still could call within their isolated groups, and when a link came in to connect the groups they could call everyone |
| 13:33.22 | Katty | irroot: not unless they're tame! |
| 13:33.33 | angryuser | b0ot, Dundi |
| 13:34.09 | irroot | nah what fun is that .... keep a fire going they stay away problem is pesky rhinos :P |
| 13:34.15 | angryuser | google it^ |
| 13:35.27 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 13:35.51 | b0ot | I take it their is no such thing as a dynamic dialplan |
| 13:36.23 | irroot | bOot can be as dynamic as you want realtime and Dundi and perhaps ENUM |
| 13:36.26 | Katty | irroot: what's the matter with rhinos? something tells me they're not too interested in your pb and jelly sammich |
| 13:36.58 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 13:37.19 | irroot | Katty they see for shit blind and have attitude ... smell fire and run for it and put it out ... dont want to be in the way camp outside there area |
| 13:38.05 | b0ot | !Dundi |
| 13:38.08 | *** part/#asterisk mameluk (~Adium@77.239.239.16) |
| 13:38.12 | leifmadsen | ~dundi |
| 13:38.13 | infobot | hmm... dundi is at http://www.dundi.com. DUNDi, an optional Asterisk component, is a distributed, decentralized peer to peer network that provides routes to PSTNs between peers on the same DUNDi network. |
| 13:38.22 | Katty | irroot: so did you sleep in the trees to avoid the rhinos? |
| 13:38.32 | b0ot | ~ENUM |
| 13:38.32 | infobot | it has been said that enum is http://www.voip-info.org/wiki-Enum |
| 13:38.34 | irroot | at the base of a baobab |
| 13:38.48 | Katty | i'm guessing that's a rather large tree |
| 13:38.49 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:38.59 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 13:41.18 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:41.18 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:41.52 | irroot | they massive indeed |
| 13:42.41 | b0ot | Dundi looks like it trys to find a way to the internet to complete calls. |
| 13:43.22 | leifmadsen | DUNDi is a question/answer protocol that you get to ask peers how to reach an end point, and the cluster returns an answer |
| 13:43.43 | leifmadsen | it provides you the data to connect to the peer (and does not actually connect you) |
| 13:43.53 | leifmadsen | ~thebook |
| 13:43.54 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 13:44.02 | leifmadsen | contains info about setting up DUNDi |
| 13:49.00 | b0ot | nice |
| 13:49.12 | b0ot | it looks like I could use DUNDi in a mobile network |
| 13:49.47 | b0ot | Any idea how long it would take the DUNDi to adjust when there is a topology change |
| 13:50.04 | *** join/#asterisk radic (~radic@dslb-178-002-216-210.pools.arcor-ip.net) |
| 13:50.36 | kaldemar | b0ot: it's a reactive protocol with caching. you can make it query every time when a number is dialed if you want to. |
| 13:51.02 | leifmadsen | b0ot: immediately as the end point that responds "registers" to the other end points |
| 13:51.10 | leifmadsen | if the end point doesn't answer, it doesn't answer |
| 13:51.18 | leifmadsen | just don't use caching |
| 13:51.29 | leifmadsen | or use it, and then you know how long it takes to adjust |
| 13:51.52 | leifmadsen | dundi doesn't care about how the data is being passed, so saying it'll work across a mobile network is kind of superfluous |
| 13:57.27 | *** join/#asterisk engrxyz (~puitpyitr@212.23.51.7) |
| 13:57.58 | b0ot | Alright well if I'm A and to dial 1234 DUNDi finds that it can go B->C->D->1234 |
| 13:58.20 | b0ot | however than the network changes and it now is A->E->D->1234 |
| 13:58.26 | b0ot | how would DUNDi handle that |
| 13:58.29 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:58.38 | b0ot | or A->C->E->D |
| 14:07.22 | kaldemar | b0ot: if will know when it goes A->E->D->1234 or A->C->E->D just like with the first query. |
| 14:07.39 | *** join/#asterisk neurosys (~neurosys@adsl-184-32-185-142.mia.bellsouth.net) |
| 14:08.43 | neurosys | *v1.6.2.11... from in the SIP header shows the user@HOST instead of the USER@ASTERISKBOX. Any ideas on why? |
| 14:16.07 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 14:16.13 | *** join/#asterisk GreatSUN (~greatsun@178-190-210-235.adsl.highway.telekom.at) |
| 14:17.52 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 14:17.52 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:19.57 | tuxx- | Hm, weird. I'm trying to reach a sippeer on my asterisk box, the call comes in through a siptrunk, and i get the following error when picking up the call: process_sdp: Insufficient information for SDP (m= not found). I tried setting the insecure value for my sippeer to port,invite. but that didnt work. Anyone got a suggestion? I got the full log with sipdebug etc here: http://pastie.org/1966333 |
| 14:20.52 | tuxx- | https://issues.asterisk.org/bug_view_page.php?bug_id=9398, this might be related, but its for asterisk 1.4.2, currently running 1.8.3.2 |
| 14:23.30 | tuxx- | on line 814 the error pops up :P |
| 14:24.37 | *** join/#asterisk BeeBuu (b71b335b@gateway/web/freenode/ip.183.27.51.91) |
| 14:25.12 | irroot | the answer is @ 711 |
| 14:25.45 | tuxx- | Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) |
| 14:25.47 | BeeBuu | how can i do mixmonitor in application queue? |
| 14:26.00 | tuxx- | so i accept gsm/ulaw/alaw and the peer only accepts ulaw? |
| 14:26.10 | tuxx- | shouldnt be a problem right, since they both ahve ulaw? :P |
| 14:26.19 | irroot | yeah |
| 14:26.55 | irroot | sorry mis read 2 lines together saw nothing :P |
| 14:28.36 | irroot | im almost done for the day hometime ill try look at it in a bit if i can |
| 14:29.08 | *** join/#asterisk fauxalliance (~fauxallia@142.163.151.196) |
| 14:29.32 | muiro | is there any way I can make asterisk take an action on a SIP or IAX2 registration? |
| 14:29.38 | tuxx- | tnx irroot |
| 14:29.57 | tuxx- | muiro: you could catch the event on AMI |
| 14:30.06 | tuxx- | and do something with it |
| 14:30.23 | muiro | is that the only way? it seems rather bulky |
| 14:30.56 | tuxx- | afaik yes |
| 14:31.04 | muiro | alright |
| 14:31.13 | muiro | I was afraid of that but I guess it confirms what I was thinking |
| 14:31.15 | tuxx- | but maybe someone in this channel has a better idea :-) |
| 14:32.02 | muiro | though, I'm not sure the AMI has an event to set custom device states. I could always shell out and do it with the console, assuming the ami listener is on the same box |
| 14:35.12 | irroot | tuxx throwing it out you have ULAW on the server set but does the phone have it in its codec list ?? |
| 14:35.18 | tuxx- | irroot: think i found it, it was a codec problem after all, fixed it by playing around with codecs on my phone. |
| 14:35.28 | irroot | beat you too it |
| 14:35.32 | tuxx- | hehe :D |
| 14:35.46 | tuxx- | tnx for the help =) |
| 14:37.18 | *** join/#asterisk fish-bulb (~qcstewart@nat/digium/x-fehumuevfalqktep) |
| 14:41.43 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:41.43 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:44.55 | *** join/#asterisk DaneoShiga (~DaneoShig@187.39.186.69) |
| 14:45.23 | DaneoShiga | someone could tell me if the "order" of the events that asterisk throws are always the same? |
| 14:45.58 | DaneoShiga | I mean, when an agent hangups a call, first comes a "AgentComplete" then an "ExtensionStatus", it's always like that? or it can happen in other order? |
| 14:55.17 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 14:56.49 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:57.51 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:58.29 | jaytee | wow, scathing article on nerdvittles about Digium and 1.8.4 |
| 14:58.37 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 14:58.49 | leifmadsen | jaytee: welcome to last week |
| 14:59.08 | jaytee | yeah, I'm just tryin to get caught up on my reading :-) |
| 14:59.17 | russellb | yawns |
| 14:59.36 | cneb3000 | jaytee: news to me - thanks for letting us know ;) |
| 14:59.42 | leifmadsen | russellb: I found some espresso in the cabinet! |
| 14:59.49 | leifmadsen | I made a double american |
| 14:59.53 | leifmadsen | double americano |
| 15:00.01 | leifmadsen | although I no speak americano |
| 15:00.22 | jaytee | bet the author gets kickbacks from PIAF |
| 15:00.34 | leifmadsen | the author builds PIAF |
| 15:00.56 | russellb | leifmadsen: nice! |
| 15:01.07 | leifmadsen | russellb: ya I'm pretty excited |
| 15:01.15 | russellb | i still have no coffee and it sucks a lot. |
| 15:03.16 | leifmadsen | dislikes |
| 15:03.48 | irroot | im with russellb STFU leifmadsen out of coffee here at office ... grumpy @ 90% 5pm so crank up the esspresso at home soon |
| 15:05.07 | leifmadsen | irroot: if I didn't find espresso though, I'd have been coffee-less all day as I have no bike and no car here at home |
| 15:05.31 | chazzam | leifmadsen: I imagine if you walked to somewhere you would be pretty awake by the time you got there |
| 15:05.48 | tuxx- | i need nicotine, but i dont have anymore cigarettes >_> |
| 15:05.55 | tuxx- | need to actually phisically go outside |
| 15:05.57 | tuxx- | and get some |
| 15:06.00 | tuxx- | *shivers* |
| 15:06.20 | leifmadsen | chazzam: possibly -- but then I'd be out of the office for over an hour just to get coffee :) |
| 15:06.22 | tuxx- | physically* |
| 15:06.26 | russellb | luckily we have a coffee shop in the building ^_^ |
| 15:06.33 | russellb | once i get off the phone, i am hitting that up. |
| 15:06.34 | leifmadsen | russellb: I'm jealous |
| 15:06.40 | irroot | applies for job @ digium |
| 15:06.44 | russellb | irroot: hired |
| 15:06.46 | leifmadsen | russellb: you're going to figuratively "tap that ass"? |
| 15:06.58 | tuxx- | hehe :-) |
| 15:07.03 | russellb | irroot: seriously, we have an opening. interested? :-) |
| 15:07.12 | leifmadsen | irroot: be interested! |
| 15:07.21 | irroot | hehe not sure that they allow me into the states ;) |
| 15:07.56 | russellb | for the guy writing t.38 gateway, remote contract work could be an option, heh |
| 15:08.01 | leifmadsen | w00t |
| 15:08.29 | russellb | attempts peer pressure in a public location :-p |
| 15:08.35 | irroot | lol ie do what im doing at the moment for me customers and digium |
| 15:08.57 | russellb | pretty much |
| 15:09.16 | *** join/#asterisk doulos1 (~bcalhoun@71-14-6-250.static.gwnt.ga.charter.com) |
| 15:09.18 | tuxx- | what kind of a job opening do you guys have? |
| 15:09.21 | tuxx- | cant find it on the website :p |
| 15:09.25 | irroot | can be a plan hey now that its winter here summer there see what im thinkin |
| 15:09.26 | russellb | Asterisk C development |
| 15:09.29 | russellb | should be on the web site |
| 15:09.39 | chazzam | http://www.digium.com/en/company/careers/ |
| 15:09.42 | chazzam | should be there |
| 15:09.56 | tuxx- | *reads* |
| 15:10.13 | chazzam | although opera apparently still only gets the first listing instead of the full list |
| 15:10.22 | chazzam | at least for me |
| 15:10.23 | russellb | should be 5 listed |
| 15:10.45 | leifmadsen | ya I got 5 here with chrome |
| 15:10.56 | chazzam | yeah, all 5 show up in chrome and firefox |
| 15:10.58 | tuxx- | same =) |
| 15:11.22 | WIMPy | sees 0 |
| 15:11.35 | russellb | stupid open source software |
| 15:11.52 | chazzam | http://digium.theresumator.com/apply/jobs/ how about there? |
| 15:12.08 | chazzam | opera sees all of the there |
| 15:12.24 | russellb | but anyway, for anyone listening, you can contact me if you're interested in either of the software positions listed. |
| 15:12.34 | WIMPy | Yes, 5. |
| 15:13.38 | mazpe | I hate that flowroute only accepts Amazon Payments |
| 15:14.35 | irroot | hehe douglas here says he will write recomendation letter ... ie no chance :P |
| 15:14.57 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.119.90) |
| 15:15.23 | leifmadsen | dougies |
| 15:16.47 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 15:17.22 | irroot | catch you tommorow off now |
| 15:17.37 | russellb | have a nice evening! |
| 15:17.40 | leifmadsen | peas out |
| 15:17.44 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 15:18.41 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 15:19.26 | irroot | http://tinyurl.com/3o5kljq Sala Gahle all |
| 15:23.39 | *** join/#asterisk bent_screwdriver (~UserNick@74.255.249.66) |
| 15:25.35 | *** join/#asterisk Diffen (~diffen@c-fc73e555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 15:28.45 | Diffen | Evning, In my asterisk 1.4 im using agents in queues so we can take queue calls on our cellphones. seems like I must have touched some setting because i can use ## for blind transfer but not #* for attendant transfer. It have been working before. When I press #* the call ends and there is nothing strange in the log. Does anyone have some sort of clue? |
| 15:29.28 | cneb3000 | Diffen: can you copy/paste the log to pastebin.com? :) |
| 15:29.42 | Diffen | cneb3000: sure hold on :D |
| 15:29.50 | cneb3000 | make sure you run this in the asterisk CLI first... "core set verbose 5" |
| 15:31.25 | Diffen | cool ill check |
| 15:34.07 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 15:39.35 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 15:41.14 | Diffen | cneb3000: should i use some syntax highlight? |
| 15:41.32 | cneb3000 | Diffen: no! no need |
| 15:41.38 | Diffen | ok ok :D:D |
| 15:43.08 | Diffen | http://pastebin.com/u0PJiZqx |
| 15:43.21 | cneb3000 | Diffen: Ok let's have a look |
| 15:44.40 | cneb3000 | Diffen: There's quite a lot there.. can you simplify what I'm looking for? |
| 15:44.45 | Diffen | yes |
| 15:45.04 | Diffen | from row 232 |
| 15:45.06 | cneb3000 | Diffen: For example.. 'An inbound call came to xxxx and I tried to transfer it by pushing 'xx' and then it failed |
| 15:45.09 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 15:47.25 | Diffen | from row 232 it starts ringing in my phone. that works fine. then i press #* somewhere around row 243-246 |
| 15:48.23 | Diffen | There isnt any error messages or anything. it just hangs up the call. in my features.conf i have the # |
| 15:48.33 | Diffen | #* for attendant transfer |
| 15:49.00 | Diffen | blindxfer => ## ; Blind transfer |
| 15:49.00 | Diffen | atxfer => #* ; Attended transfer |
| 15:49.17 | Diffen | and a blind transfer works perfectly... strange :) |
| 15:49.49 | cneb3000 | Diffen: Have you made any changes at all recently? |
| 15:50.00 | Diffen | no not that i can remember... |
| 15:50.05 | Diffen | so no |
| 15:50.53 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 15:51.20 | cneb3000 | Diffen: hmm..! I want to help but cant spot anything in the logs myself..! |
| 15:51.54 | Diffen | np i will remove the queue and see if it works. then restart my asteirsk |
| 15:52.13 | cneb3000 | try changing to something different. *2 for example.? |
| 15:52.32 | Diffen | yes will do that |
| 15:53.29 | *** join/#asterisk coppice (~chatzilla@122.225.226.140) |
| 15:54.27 | cneb3000 | Diffen: can you copy paste your 'dial-SIP' extension to paste bin? |
| 15:58.15 | atan | What does into the allow= field within sip.con to enable G.711? |
| 15:58.27 | russellb | allow=ulaw,alaw |
| 15:58.29 | Cadey | uLaw? |
| 15:58.53 | atan | Ahh yes. |
| 15:58.55 | atan | Thank you. |
| 16:02.53 | leifmadsen | well there are 2 flavors of G.711 (ulaw and alaw) |
| 16:03.04 | leifmadsen | so what russellb said is correct |
| 16:03.15 | atan | US and Europe I think |
| 16:03.17 | jaytee | ulaw tastes better than alaw |
| 16:03.20 | atan | I totally just forgot is all. :-) |
| 16:03.30 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 16:03.43 | atan | It uses far more bandwidth than gsm. Hmm. |
| 16:03.44 | *** join/#asterisk highvoltz (rogers@bling.bling.org) |
| 16:03.51 | leifmadsen | 64kbit/s yes |
| 16:03.57 | leifmadsen | the quality is also significantly better |
| 16:04.05 | leifmadsen | gsm sounds awful |
| 16:04.43 | jaytee | and tastes lousy compared to G.711 flavors :-) |
| 16:04.48 | atan | Well the kicker is it's being sent over a cellular connection. I wasn't sure if it would do well with a higher codec but it seems fine :D |
| 16:06.13 | jaytee | dl1.digium.com is slooooooow. 190-200 kbps download |
| 16:06.29 | russellb | blames leifmadsen |
| 16:06.40 | leifmadsen | has no control over that site :) |
| 16:06.58 | jaytee | and I've got a 50down 10up circuit so it's gotta be the server throttling |
| 16:07.15 | leifmadsen | you get what you pay for? :) |
| 16:07.44 | jaytee | hahaha, guess so. glad I'm a patient guy. probably just a busy day for downloads |
| 16:07.48 | coppice | leifmadsen: if GSM sounds awful, you must be using a broken one. its somewhat poorer than alaw and ulaw, although those aren't particularly wonderful to begin with |
| 16:08.11 | atan | wonders what a broken one is. |
| 16:08.23 | atan | See O dodm |
| 16:08.25 | jaytee | we need a "Vader" codec |
| 16:08.39 | coppice | there are plenty of broken codecs implementations around |
| 16:08.42 | jaytee | normal voice in > Vader voice out |
| 16:08.43 | atan | Err. I didn't notice much trouble with the gsm codec. Nobody said anything to me about the call quality being bad. |
| 16:08.45 | leifmadsen | coppice: right, so any loss in quality over ulaw (which is already not awesome) is worse |
| 16:08.51 | russellb | jaytee: lpc10? |
| 16:08.56 | leifmadsen | I stick with my original statement |
| 16:09.20 | _Corey_ | anyone know if any cell carriers are doing anything better than gsm? I got a new bluetooth headset the other day that touts wideband audio and couldn't help wondering why... |
| 16:09.32 | jaytee | wonders what to have for lunch |
| 16:09.38 | leifmadsen | _Corey_: some are testing higher quality codecs, yes |
| 16:09.39 | WIMPy | coppice: Yes, including the iPhone, I think. |
| 16:10.04 | _Corey_ | aha, well i'm all anticipation now |
| 16:10.08 | coppice | _Corey_: a few networks now support wideband over cellular |
| 16:11.05 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 16:12.07 | coppice | most early 3G phones had wideband codecs. when the networks blocked its use, the phones dropped it. they are slowly putting it back |
| 16:12.10 | atan | Are there any test phone numbers which play high quality audio? |
| 16:12.39 | coppice | russellb: even LPC10 doesn't have to sound as bad as the broken codec in * |
| 16:12.40 | _Corey_ | hmm, i guess we'll notice it first between users of the same cell network |
| 16:13.45 | coppice | _Corey_: I don't think the networks are even attempting to do wideband with other networks |
| 16:14.08 | _Corey_ | yeah, I wouldn't expect that to happen anytime soon |
| 16:14.47 | atan | http://www.headset-plus.com/plantronics-cs510-wireless-headset-system-p-1156.html wowsers, $99,999.00 must be a good headset! |
| 16:18.49 | coppice | atan: the dollar is starting to gain some of it value back, so it might be cheaper in Sep when its launched |
| 16:19.20 | *** join/#asterisk pdtpatrick (~pdtpatric@mainstwan.farheap.com) |
| 16:22.43 | atan | The comedy :D it kills |
| 16:23.33 | DaneoShiga | someone could tell me if the "order" of the events that asterisk throws are always the same? |
| 16:23.35 | DaneoShiga | I mean, when an agent hangups a call, first comes a "AgentComplete" then an "ExtensionStatus", it's always like that? or it can happen in other order? |
| 16:24.32 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 16:31.32 | *** join/#asterisk sunfone (~jeff@c-67-184-36-158.hsd1.il.comcast.net) |
| 16:34.33 | *** join/#asterisk \DSAFEW\ (~\DSAFEW\@ip68-3-53-165.ph.ph.cox.net) |
| 16:34.35 | *** join/#asterisk afink_ (~afink@204.26.87.226) |
| 16:40.01 | *** join/#asterisk drudge` (~drudge@unaffiliated/drudge/x-837452) |
| 16:41.00 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 16:42.17 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 16:45.03 | *** join/#asterisk n0tk (debian-tor@gateway/tor-sasl/n0tk) |
| 17:03.07 | el3slave | you guys have any preference on headsets for sip softphones? |
| 17:03.36 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 17:04.06 | WIMPy | Plantronics/GN |
| 17:04.58 | el3slave | thanks WIMPy |
| 17:05.28 | el3slave | think blue tooth would cause issues? |
| 17:05.52 | WIMPy | I haven't tried that. |
| 17:06.31 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 17:06.54 | WIMPy | But it should work just as any other audio device. |
| 17:07.02 | el3slave | yea |
| 17:07.06 | el3slave | thanks sir |
| 17:10.15 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 17:15.33 | wesphillips | does anyone know about the jtterbuffer in asterisk? I am trying to enable a buffer on a SIP <-> SIP bridge, and I added the jbenable=yes and jbforce=yes to the sip.conf, but I get no message on the CLI, and no logfile created(I have jblog=yes as well). I am running 1.6.2.9 |
| 17:16.38 | *** join/#asterisk cusco (~tralala@a89-152-96-250.cpe.netcabo.pt) |
| 17:16.43 | cusco | hi |
| 17:17.13 | WIMPy | I've never trid to look at that, but are you sure, your Asterisk is in the media path at all? |
| 17:17.24 | cusco | im having trouble configuring asterisk 1.8.4.1 --with-ssl |
| 17:17.27 | cusco | checking for ssl2_connect in -lssl... no |
| 17:17.29 | wesphillips | it is. I have canreinvite=no |
| 17:17.59 | WIMPy | And directmedia? |
| 17:18.47 | wesphillips | I also know that it is in the path for two reasons, one is that I am coverting from G.729 20 byte payloads to G.729 60 byte payloads and I am also doing a wireshark capture on both sides of the Asterisk bridge. |
| 17:19.16 | WIMPy | ok |
| 17:19.34 | leifmadsen | cusco: sounds like you don't have the development files asterisk needs installed |
| 17:19.45 | cusco | I have libssl-dev (debian) |
| 17:20.28 | cusco | folks in #debian tell me that ssl2 is dead |
| 17:20.45 | leifmadsen | you probably need the changes that are in asterisk 1.8 |
| 17:20.49 | leifmadsen | (the branch directly) |
| 17:21.51 | wesphillips | i tried changing the reverse direction from 60 to 20 byte payloads, and I noticed that asterisk was spitting out 3 voice packets at a time(I was seeing a 60 ms interpacket delay delta and then two zero interpacket delay deltas) , so that also confirms that there is no buffer. |
| 17:22.04 | *** join/#asterisk cusco_ (~tralala@a89-152-96-250.cpe.netcabo.pt) |
| 17:22.07 | cusco_ | sorry |
| 17:22.12 | cusco_ | connection felll |
| 17:22.18 | cusco_ | I might not have catched your last messages |
| 17:24.30 | cusco_ | "18:23 < stew> cusco_: according to the debian changelog, the debian patch to fix this problem was accepted upstream and is included in 1.8.4" |
| 17:24.48 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 17:29.14 | *** join/#asterisk sircolin (~sircolin@my83-216-68-241.mynow.co.uk) |
| 17:29.47 | *** part/#asterisk sircolin (~sircolin@my83-216-68-241.mynow.co.uk) |
| 17:30.00 | *** join/#asterisk sircolin (~sircolin@my83-216-68-241.mynow.co.uk) |
| 17:30.37 | cusco | leifmadsen: I actually see that in the notes... from debian package: http://paste.debian.net/117861/ |
| 17:30.56 | cusco | so I should be ok to mimic that patch by altering configure.ac |
| 17:31.18 | leifmadsen | is it *actually* included in 1.8.4? I don't think so. |
| 17:31.32 | leifmadsen | pretty sure I committed those changes, and they were done recently (post 1.8.4-rc1) |
| 17:32.22 | *** join/#asterisk cyphorious (~cyphoriou@chello062178189196.2.15.tuwien.teleweb.at) |
| 17:33.14 | cusco | so Im using 1.8.4.1 it should have that |
| 17:33.18 | cusco | goes check configure.ac |
| 17:33.24 | leifmadsen | cusco: why do you think that? |
| 17:33.26 | leifmadsen | did you look at the log file? |
| 17:33.32 | leifmadsen | s/log file/ChangeLog file/ |
| 17:33.42 | leifmadsen | 1.8.4.1 has only certain fixes |
| 17:33.45 | cusco | ow... |
| 17:33.53 | leifmadsen | like I said, you more than likely need the latest 1.8 branch |
| 17:33.57 | cusco | goes looking at changelog first |
| 17:34.51 | leifmadsen | anyways I'm telling you what you need |
| 17:37.30 | cusco | yes, the latest branch |
| 17:37.31 | el3slave | really enjoying the definitive guide, leifmadsen, thanks! |
| 17:38.10 | cusco | non-related: how does one buy the definitive guide? |
| 17:38.15 | cusco | :p |
| 17:38.17 | leifmadsen | el3slave: glad you're enjoying it |
| 17:38.26 | leifmadsen | cusco: amazon.com makes it easy |
| 17:38.27 | cusco | I could actually do some reading while on public transport |
| 17:38.33 | cusco | they're expensive |
| 17:38.54 | cneb3000 | Cusco: Buy the kindle version |
| 17:39.02 | cneb3000 | Cusco: Actually.. fiirst buy a kindle... :P |
| 17:39.02 | cusco | cneb3000: why? what is it? |
| 17:39.06 | cusco | ah |
| 17:39.06 | cusco | ! |
| 17:39.07 | leifmadsen | https://issues.asterisk.org/view.php?id=19138, https://issues.asterisk.org/view.php?id=19095 <-- cusco |
| 17:39.12 | cusco | thats damn expensive too! |
| 17:39.19 | jaytee | I <3 my Kindle |
| 17:39.26 | cusco | click click... |
| 17:39.28 | leifmadsen | cusco: buying things requires the exchange of cash unfortunately |
| 17:39.47 | cusco | chash should be foss |
| 17:39.55 | cneb3000 | Jaytee: I found it's made me read more... Sad to think it's mainly because i'm reading it off tech as opposed to paper eh? |
| 17:40.05 | leifmadsen | cusco: see bitcoin |
| 17:40.07 | leifmadsen | ~bitcoin |
| 17:40.08 | jaytee | according to Jean Luc Picard, in the 24th century they don't use cash anymore :-) |
| 17:40.35 | cusco | "Use SSLv23_client_method instead of old SSLv2 only." OK |
| 17:40.44 | cusco | leifmadsen: I did and had a laugh with it |
| 17:40.47 | *** join/#asterisk sircolin (~sircolin@my83-216-68-241.mynow.co.uk) |
| 17:40.50 | *** part/#asterisk sircolin (~sircolin@my83-216-68-241.mynow.co.uk) |
| 17:40.51 | cusco | is the book available trough bitcoin? |
| 17:40.58 | cneb3000 | ^^ haha |
| 17:40.59 | leifmadsen | not from me |
| 17:41.03 | cusco | :P |
| 17:41.14 | jaytee | cneb3000, I just like the idea of having 5 or 6 good technical books like Asterisk: The Definitive Guide, Asterisk Cookbook, etc. and not have to carry 20 pounds of paper around. |
| 17:41.22 | leifmadsen | jaytee: +1 |
| 17:41.27 | jaytee | although I also have the print version |
| 17:41.30 | jaytee | of each |
| 17:41.31 | leifmadsen | lol |
| 17:41.35 | cneb3000 | jaytee: amen |
| 17:41.39 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:41.53 | leifmadsen | I think it would have been faster to download this Fedora ISO image from the internet than across my LAN for some reason |
| 17:42.08 | leifmadsen | pscp must add a bunch of overhead |
| 17:43.14 | *** join/#asterisk sircolin (~sircolin@my83-216-68-241.mynow.co.uk) |
| 17:43.15 | jaytee | I usually get solid throughput with WinSCP |
| 17:43.23 | leifmadsen | ya |
| 17:43.29 | leifmadsen | odd, oh well |
| 17:43.30 | leifmadsen | no rush |
| 17:44.02 | *** part/#asterisk sircolin (~sircolin@my83-216-68-241.mynow.co.uk) |
| 17:44.17 | jaytee | Fedora, huh? Jared trying to convert you from CentOS? |
| 17:44.35 | leifmadsen | Fedora and CentOS are two totally separate things |
| 17:44.45 | leifmadsen | I wouldn't deploy Fedora onto an Asterisk server |
| 17:44.51 | leifmadsen | I would certainly try it out on my desktop though |
| 17:45.00 | leifmadsen | so there is no relationship there for converting |
| 17:45.02 | jaytee | ah, yeah. it makes a great desktop |
| 17:45.22 | jaytee | you could run * on it if you wanted to |
| 17:45.23 | leifmadsen | I use ubuntu server edition and centOS for server deployments |
| 17:45.28 | cusco | I find that debian makes great everything |
| 17:45.33 | leifmadsen | I could, and I might for development purposes |
| 17:45.36 | cusco | from laptop to servers |
| 17:45.44 | cneb3000 | oh no. a distro discussion! |
| 17:45.47 | cneb3000 | runs and hides |
| 17:45.47 | cusco | hehehe |
| 17:45.48 | cneb3000 | :) |
| 17:45.50 | leifmadsen | ya it's stupid |
| 17:45.53 | leifmadsen | I use what works |
| 17:45.56 | cusco | true |
| 17:46.02 | jaytee | yep |
| 17:46.09 | leifmadsen | so... about that asterisk stuff |
| 17:46.11 | cusco | tho I might add, debian has DFSG that stands for your rights regarding software |
| 17:46.23 | cusco | (I'm finished) |
| 17:46.28 | cneb3000 | lol |
| 17:47.45 | cneb3000 | have "pirates of silicon valley" running in the background.. good movie :) |
| 17:48.00 | leifmadsen | I wouldn't go that far |
| 17:48.06 | cusco | never watched it |
| 17:48.13 | cusco | I enjoyed antitrust |
| 17:48.33 | cusco | with that ryan something actor |
| 17:48.46 | cusco | its nice how they paint microsoft |
| 17:48.47 | leifmadsen | Reynolds? |
| 17:48.47 | cusco | :P |
| 17:48.59 | cusco | no... ryan phillip I think... |
| 17:49.13 | cusco | http://www.imdb.com/title/tt0218817/ |
| 17:50.10 | cneb3000 | Ha. Do you find when you read description of a 'computer film' that they tend to sound tacky |
| 17:50.44 | cusco | what is tacky? like dodgy? |
| 17:50.51 | cneb3000 | hm... |
| 17:50.58 | cusco | they often omit the specifics ... |
| 17:51.00 | cneb3000 | how to explain tacky? |
| 17:51.07 | cusco | is it on dict.org ? |
| 17:51.20 | cneb3000 | 'Showing poor taste and quality: "his tacky decor". |
| 17:51.33 | cneb3000 | its a british thing... haha |
| 17:52.34 | *** join/#asterisk m_tadeu (~quassel@89.180.67.125) |
| 17:54.10 | m_tadeu | hi...I'm using ami to set a cdr field, and asterisk replies with success...but the field doesn't seem to change. what can I be doing wrong? |
| 17:57.09 | *** join/#asterisk Tim_Toady (~moi@188.4.51.59.dsl.dyn.forthnet.gr) |
| 17:57.18 | m_tadeu | I'm using the SetVar action, btw |
| 17:57.28 | cusco | shouldn't it be Set() ? |
| 17:57.46 | cusco | leifmadsen: latest 1.8 branche works, thanks again :p |
| 17:58.27 | m_tadeu | cusco: in the reference says SetVar |
| 17:59.01 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 17:59.07 | leifmadsen | m_tadeu: which reference? |
| 17:59.10 | leifmadsen | SetVar() is crazy old |
| 17:59.18 | leifmadsen | it's been Set() since like.... 1.2? |
| 17:59.28 | m_tadeu | https://wiki.asterisk.org/wiki/display/AST/ManagerAction_Setvar |
| 17:59.33 | leifmadsen | oh you're talking the AMI setvar |
| 17:59.36 | leifmadsen | sorry, I misunderstood |
| 18:03.08 | cusco | so did I |
| 18:03.18 | m_tadeu | in deed...I'm setting the variable: CDR(answered)...which doesn't exist in the original CDR...but should't represent a problem, since I'm also using a CDR(queued) without a problem(but setting this in an agi script) |
| 18:04.04 | leifmadsen | I don't think you can change any fields except userfield |
| 18:05.37 | cusco | I use Set(CDR(Userfield)=bla); and Set(CDR(accountcode)=Outbound912327540) |
| 18:05.56 | m_tadeu | yes I'm also settin the account code somewhere in an agi script |
| 18:06.38 | m_tadeu | and also a custom one CDR(queued),, which marks the time the caller entered a queue |
| 18:08.49 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 18:11.31 | *** join/#asterisk sircolin (~sircolin@my83-216-68-241.mynow.co.uk) |
| 18:13.21 | cusco | and that is working, you say |
| 18:14.00 | diatonic | So, if they ever make a hollywood movie about the free software movement, who plays richard stallman? Maybe zach galifianakis? |
| 18:14.04 | cusco | are you sure CDR(answered) is not set? |
| 18:14.12 | cusco | have you checked with dumpchan() or so? |
| 18:14.58 | cusco | diatonic: I don't know that guy, but I'm sure they could ask stallman himself |
| 18:14.59 | *** join/#asterisk reckio1 (~reckio@189-69-16-26.dsl.telesp.net.br) |
| 18:15.04 | cusco | Im sure he would love to have a say in it |
| 18:15.18 | cusco | and propagate some more religion trough new channels |
| 18:15.21 | cusco | hehe |
| 18:15.28 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:16.35 | reckio1 | does anyone integrated spa9XX phones with ad 2008 |
| 18:16.37 | reckio1 | ? |
| 18:17.19 | m_tadeu | cusco: the value should show up in the CDR record, I guess |
| 18:18.19 | cusco | I guess.. I would use dumpchan() just to be sure |
| 18:18.30 | cusco | and check verbose mode |
| 18:18.35 | cusco | or debug mode |
| 18:18.42 | cusco | to see what happens when you use setvar |
| 18:20.04 | m_tadeu | it's the first time I hear about dumpchan....how do I use it? |
| 18:20.17 | leifmadsen | just use it from the dialplan and look at the output |
| 18:21.19 | m_tadeu | so it will dump the info at that time, right? |
| 18:21.56 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 18:23.49 | pdtpatrick | Can you guys please tell me what's the advantage to 1.8 asterisk vs 1.6 .. we currently have 1.6 and im looking to upgrade. Or what is the big difference between AsteriskNow vs Asterisk .. thanks in advance |
| 18:24.42 | *** join/#asterisk lesouvage (~lesouvage@62.140.137.156) |
| 18:25.00 | fish-bulb | pdtpatrick: 1.8 is a more recent version and includes new features. 1.6.2 is no longer receiving bug fixes, so not the best idea to use for a new install |
| 18:25.13 | p3nguin | 1.8 is a Long-Term Support branch, where 1.6.0, 1.6.1, and 1.6.2 are not. AsteriskNOW is a complete CentOS Linux distribution with Asterisk preinstalled and with FreePBX and the Asterisk GUI available. |
| 18:25.16 | fish-bulb | and you will only get a little less than a year of security fixes too |
| 18:25.25 | fish-bulb | annnd... what p3nguin said |
| 18:27.34 | m_tadeu | I can't see the dumpchan, but something that worth notice in the console came up..."app_macro.c:304 _macro_exec: No such context 'macro-answered' for macro 'answered'"...well, answered is the cdr value I'm trying to set |
| 18:27.42 | *** join/#asterisk wesphillips (~wphill04@137.237.233.124) |
| 18:28.28 | pdtpatrick | oh i see ... thanks guys! |
| 18:33.53 | pdtpatrick | is there a way to source remote files on asterisk? for instance i have something in another server that i'd like to reference in my current extensions.conf or sip.conf |
| 18:33.58 | pdtpatrick | is that possible and how ? |
| 18:34.00 | cusco | m_tadeu: that is regarding a macro. You're somehow triggering a macro that doesn't exist? Or a context that does not exist |
| 18:34.07 | cusco | check the course on cli |
| 18:34.36 | cusco | pdtpatrick: I use sftp to place .call files... |
| 18:35.10 | _Corey_ | pdtpatrick: One technique is to use NFS to share a config folder |
| 18:35.21 | pdtpatrick | that would work just fine? |
| 18:35.40 | _Corey_ | just make sure your machine-specific stuff isn't shared in a way that will cause problems |
| 18:36.16 | _Corey_ | I've seen people do like an "asterisk" and "asterisk-local" where all the machine specific stuff is #included from the asterisk-local |
| 18:36.37 | leifmadsen | _Corey_: ya I've done that |
| 18:36.55 | pdtpatrick | oh nice.. let me look into that. The best way to reload sip is .. sip reload ? |
| 18:36.59 | pdtpatrick | from cli ? |
| 18:37.05 | _Corey_ | It's handy for clustered stuff that can't do everything realtime |
| 18:37.50 | _Corey_ | or 'asterisk -rx "sip reload" ' |
| 18:39.40 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 18:45.15 | pdtpatrick | thanks! |
| 18:51.28 | *** join/#asterisk chasing`Sol (~cS@smtp.master-zone.net) |
| 18:52.06 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 18:52.16 | *** join/#asterisk chasing`Sol (~cS@smtp.master-zone.net) |
| 18:54.32 | *** join/#asterisk jpsharp (~jsharp@74-95-145-86-Naples.hfc.comcastbusiness.net) |
| 19:06.15 | ChannelZ | Well, there goes SFA |
| 19:06.29 | justdave | in the iax2 netstats, which direction is which side of the table? |
| 19:06.34 | justdave | it's labeled LOCAL and REMOTE |
| 19:06.52 | justdave | inbound and outbound would probably be more useful :) |
| 19:07.02 | justdave | unless I'm misundstanding what it's representing |
| 19:10.12 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 19:10.29 | leifmadsen | ChannelZ: eh? |
| 19:11.28 | sezuan | how suprising, Skype for Asterisk will be discontinued :( |
| 19:12.00 | ChannelZ | Skype is kicking SFA to the curb. No new sales after July, though they 'promise' to make it still work for 2 years |
| 19:12.00 | justdave | Microsoft sells a VoIP product that they don't want competition for? |
| 19:12.23 | tzanger | never did use it |
| 19:13.09 | ChannelZ | I wonder if they will ultimately get rid of their own SIP for Skype or whatever it is |
| 19:13.40 | cusco | so they bought it? |
| 19:13.48 | cusco | last I read it was not finite |
| 19:15.28 | _Corey_ | ouch, I just got Rod's e-mail |
| 19:15.50 | _Corey_ | I literally have like 3 customers who want it... bah |
| 19:16.09 | cusco | skype4asterisk? |
| 19:18.38 | jaytee | Microsoft manage to put the nail in yet another product |
| 19:19.22 | jaytee | s/manage/manages |
| 19:20.06 | _Corey_ | I'm just holding my breath for them to rename it "bing communicator" or some bs |
| 19:23.52 | ChannelZ | Microsoft Messenger For Voice Communications 2011 |
| 19:25.46 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 19:26.21 | jaytee | Microsoft Anal Sphincter for Voice Crapification 2011 |
| 19:27.22 | _Corey_ | As long as there's a talking paperclip, I'm ditching Asterisk for it |
| 19:28.02 | _Corey_ | :) |
| 19:29.07 | paulc | Haha did you see The Office the other day where Darryl called up asking how to re-enable Clippy? Made me laugh :-) |
| 19:29.14 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 19:29.35 | _Corey_ | that's hilarious |
| 19:34.55 | QuantumSchema | Good afternoon all! |
| 19:35.53 | QuantumSchema | I'm tripping over a small (kind of) part with agents, queues, and calls. |
| 19:36.00 | *** join/#asterisk b0ot (~tom@147.177.42.173) |
| 19:36.10 | b0ot | !book |
| 19:36.15 | b0ot | ~book |
| 19:36.15 | infobot | For more information about the Asterisk book, see ~thebook |
| 19:36.19 | b0ot | !thebook |
| 19:36.23 | b0ot | ~thebook |
| 19:36.23 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 19:36.26 | ChannelZ | really? |
| 19:36.36 | QuantumSchema | And I'm not quite sure on how the queue app is monitoring the usage of lines. |
| 19:36.51 | ChannelZ | By default it sort of doesn't |
| 19:36.59 | b0ot | ~thebook |
| 19:36.59 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 19:37.10 | b0ot | see ~thebook |
| 19:38.12 | QuantumSchema | Right... it's doing a great job with the whole "ringinuse" part with incomming calls. |
| 19:38.24 | ChannelZ | It tries a channel and if the channel accepts the call, off it goes |
| 19:38.43 | leifmadsen | device state is used to determine if a member of the queue is free |
| 19:39.06 | ChannelZ | If your agent's phones have call waiting turned on, they will accept multiple calls |
| 19:39.14 | leifmadsen | not necessarily |
| 19:39.28 | QuantumSchema | I agree with leif. |
| 19:39.29 | leifmadsen | it can if the device state doesn't report as InUse when a single line is in user |
| 19:39.39 | leifmadsen | s/user/use |
| 19:39.50 | QuantumSchema | Wait wait.... are we talking about inbound or outbound specifically here? |
| 19:39.54 | leifmadsen | but that would be dependent upon ringinuse={yes,no} |
| 19:40.05 | leifmadsen | inbound or outbound what? |
| 19:40.11 | leifmadsen | I'm just talking channels and members |
| 19:40.35 | QuantumSchema | Well, it knows about inbound calls and agents on devices that aren't reporting device state. |
| 19:40.50 | leifmadsen | kind of |
| 19:40.52 | QuantumSchema | That is the part I'm thinking is in the queue app... |
| 19:41.03 | leifmadsen | only SIP channels can report device state accurately |
| 19:41.04 | QuantumSchema | It's just remembering "AGENTCONNECT". |
| 19:41.18 | leifmadsen | the rest is a crap shoot |
| 19:41.23 | QuantumSchema | LoL That's what I was thinking. |
| 19:41.37 | QuantumSchema | So what I wanted to try and get to was using hints... if that's the right path. |
| 19:41.40 | leifmadsen | I only use SIP channels with app_queue because they are the most reliable |
| 19:41.48 | leifmadsen | hints just depend on device state |
| 19:41.58 | leifmadsen | it's the device state that you need configured and setup correctly |
| 19:42.04 | leifmadsen | and to do that, you need to use SIP end points |
| 19:42.23 | QuantumSchema | I wanted to make hints that correspond the agents channel... like "ExternalPBX/${EXTEN}" |
| 19:43.14 | QuantumSchema | Then in the dialplan, if something goes to that extension (in the trunk dialplans), it sets the hint to IN USE. Would that work? |
| 19:43.34 | leifmadsen | you can setup custom hints, sure |
| 19:43.37 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 19:43.49 | leifmadsen | you need to use DEVICE_STATE() for that |
| 19:44.56 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 19:49.51 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 19:57.11 | *** join/#asterisk LedHed (~LedHed@static-74-45-162-66.dr01.pasn.ca.frontiernet.net) |
| 19:58.07 | LedHed | how many concurrent calls can a T1/PRI support? (Not SIP Handoff) |
| 19:58.26 | _Corey_ | 24/23 |
| 19:58.36 | _Corey_ | (respectively) |
| 19:58.50 | leifmadsen | _Corey_: I like your answer :) |
| 19:59.00 | _Corey_ | it's accurate :) |
| 19:59.08 | leifmadsen | that's why I like it |
| 19:59.15 | LedHed | I know theres 24/23 channels but my ATT rep keeps saying the I can only get 17 with g711. |
| 19:59.30 | leifmadsen | if you're using SIP over a T1 data link, that may be true |
| 19:59.33 | p3nguin | That's probably a data T1 circuit. |
| 19:59.34 | _Corey_ | well, you did say "not sip handoff" |
| 19:59.38 | LedHed | but when the calls leave Asterisk over the PRI its not g711 correct |
| 19:59.40 | leifmadsen | there is IP overhead, etc. |
| 19:59.54 | LedHed | _Corey_, correct, NOT SIP Handoff |
| 20:00.07 | p3nguin | If you're running a voice T1, you have 24 channels. |
| 20:00.27 | *** join/#asterisk fofware (~fabian@wdctf.siup.gov.ar) |
| 20:00.28 | _Corey_ | if you have an actual PRI, what they're saying doesn't make sense |
| 20:00.40 | LedHed | so when using a PRI card with Asterisk, what format are the calls in when they leave the server? |
| 20:01.02 | LedHed | _Corey_, I dont have a T1/PRI yet. I'm ordering one. |
| 20:01.11 | LedHed | so I should be asking for a Voice T1 |
| 20:01.22 | p3nguin | unless you want a data T1 |
| 20:01.50 | _Corey_ | If you've bought a PRI card, order a PRI |
| 20:02.08 | LedHed | p3nguin, I guess thats where the ATT rep is confused. I Said Asterisk, so I he must assume because its VoIP that its data. |
| 20:02.28 | LedHed | _Corey_, I havent ordered hardware yet either. |
| 20:02.32 | _Corey_ | ah |
| 20:02.37 | *** join/#asterisk oliver1 (~oliver@manz-590eeef7.pool.mediaWays.net) |
| 20:02.39 | p3nguin | If you have a data T1, you'll be converting it to Ethernet somewhere along the line, and then all your voice traffic will be in one of the standard VoIP protocols. |
| 20:03.38 | LedHed | ok, so for a Data T1, the calls would leave as SIP, and for a Voice T1, they leave Analog? |
| 20:03.47 | p3nguin | correct |
| 20:04.15 | _Corey_ | I'd be surprised if ATT does SIP well... anyone use them? |
| 20:04.17 | LedHed | and if using a Voice T1, I can make 23+/- calls (1 per channel) |
| 20:04.19 | keith4 | don't tell them it's Asterisk. just tell them it's a PBX |
| 20:04.38 | p3nguin | also correct |
| 20:04.43 | LedHed | and with a Data T1 (using g711) I can get 17+/- |
| 20:04.51 | p3nguin | or more if you use another codec. |
| 20:04.57 | LedHed | p3nguin, right |
| 20:05.13 | LedHed | Thanks. You've helped a lot. |
| 20:05.17 | p3nguin | The data T1 is 1.544 Mbps, so you can do the math on that. |
| 20:05.21 | keith4 | but if you get a data T1... would you still even want a T1 card for the asterisk box? |
| 20:05.36 | p3nguin | You wouldn't need it. |
| 20:05.40 | keith4 | (e.g., would you be using the * box as your network router?) |
| 20:05.51 | LedHed | no |
| 20:06.07 | p3nguin | I think most people would use some other type of network appliance for that. |
| 20:06.12 | p3nguin | I know I would. |
| 20:06.13 | keith4 | i hope so ;-) |
| 20:06.14 | LedHed | It would hit a ATT Managed router (Cisco 2811) then hit Asterisk |
| 20:06.28 | keith4 | does AT&T even offer sip service? or was he assuming you were using some itsp? |
| 20:06.32 | LedHed | yes |
| 20:06.46 | LedHed | ATT does offer SIP |
| 20:06.58 | p3nguin | The 2811 would have the appropriate WIC and then your Asterisk box would talk Ethernet to it. |
| 20:07.01 | _Corey_ | LedHed: what part of the country are you in? |
| 20:07.09 | keith4 | interesting. did not know that |
| 20:07.09 | LedHed | Cali |
| 20:07.19 | keith4 | ah, there it is: http://www.business.att.com/enterprise/Service/voice-services/voip/sip-trunking/ |
| 20:07.35 | _Corey_ | yeah, I've not used them in california but in some other markets it's like a miracle they can get a pri working |
| 20:07.48 | LedHed | lol |
| 20:07.55 | LedHed | ATT is better than the alternative. |
| 20:08.02 | LedHed | In my case thats Frontier |
| 20:08.12 | LedHed | which is a Rural Telecom |
| 20:08.38 | p3nguin | If you're using AT&T as your data T1 carrier, you can then choose any of the zillions of ITSPs out there for SIP services. |
| 20:13.00 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 20:13.02 | p3nguin | I'm not sure if a business class DSL circuit would be a better option for you or not. I know I would never consider a T1 before checking the other available services simply based on the price. |
| 20:13.03 | schmidts | good evening |
| 20:13.14 | *** join/#asterisk flyman_ (~flyman_@chello084114147007.7.15.vie.surfer.at) |
| 20:16.11 | citywok | Frontier isn't just rural telecom, they recently bought Verizon's landline biz |
| 20:16.34 | p3nguin | WHAT?! |
| 20:16.41 | citywok | yea... like a year ago... lol |
| 20:16.58 | p3nguin | Now Frontiernet is going to be replacing Verizon Landline? |
| 20:17.29 | p3nguin | Frontier has never had a good reputation for quality OR service. |
| 20:17.33 | _Corey_ | only some markets i think |
| 20:17.44 | citywok | ah, they did in my market (redmond, wa) |
| 20:18.24 | p3nguin | In most places around here they are laughed at. |
| 20:18.25 | _Corey_ | it was mostly rural areas, actually |
| 20:18.35 | _Corey_ | i'm surprised redmond was in the mix |
| 20:18.54 | _Corey_ | i remember there was a lot of fuss in west virginia about it |
| 20:19.07 | _Corey_ | the verizon union was PISSED |
| 20:19.37 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 20:22.31 | *** join/#asterisk r0d3nt (r0d3nt@foster.stonedcoder.org) |
| 20:23.37 | LedHed | All I know is our current service with Frontier is terrible. Phone is sketchy at best. and Data gets routed to NY then back to CA. Makes for very laggy internet access. |
| 20:25.38 | citywok | we are also giving up on using SIP for our primary outbound call traffic, we're going back to an old fashioned T1. Too many issues with sip carriers. |
| 20:29.32 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3a1701d.cpe.net.cable.rogers.com) |
| 20:29.50 | p3nguin | I guess a regular old voice line is a lot more reliable than several ISPs. |
| 20:30.43 | WIMPy | Yes, but then voice lines are usually supplied over DSL these days :-( |
| 20:30.58 | oliver1 | Hi there. I've tried to install a Asterisk Now-Version. But the boot-procedure stops with "NET: Registered protocol family 2". I dont know what it is. Anyone can help me please? |
| 20:35.36 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 20:37.50 | b0ot | are there any respected asterisk certifications? |
| 20:38.26 | WiretapWork | oliver1, hit enter |
| 20:38.35 | WiretapWork | it may actually be fully booted |
| 20:38.41 | WIMPy | b0ot: dcap? |
| 20:39.19 | WiretapWork | unless that is before initscripts are running, in which case it sounds like you might have a corrupted install or dodgy hardware |
| 20:39.38 | oliver1 | WiretapWork: I hit enter, but it does not work... |
| 20:39.51 | WiretapWork | oliver1, is this before the initscripts run? |
| 20:40.16 | *** join/#asterisk bent_screwdriver (~UserNick@74.255.249.66) |
| 20:40.24 | oliver1 | WiretapWork: initscript is the install options with a blue background? |
| 20:40.32 | WiretapWork | no |
| 20:40.35 | keith4 | yikes |
| 20:40.54 | WiretapWork | the initscript is where it goes "starting service <service name here>" and you get either a green "OK" or red "FAIL" on the right |
| 20:41.02 | b0ot | WIMPy, how is a dcap or dcaa considered as compared to a CCNP-Voice or CCNA-Voice |
| 20:41.28 | WiretapWork | oliver1, I should probably say that attempting to get going with Asterisk with _no_ working knowledge of linux is probably going to be extremely hard |
| 20:41.34 | oliver1 | WiretapWork: no, these part I have not seen yet |
| 20:41.45 | WIMPy | b0ot: I'n not in to that cert stuff. |
| 20:42.13 | Diffen | cneb3000: there? |
| 20:42.31 | oliver1 | WiretapWork: oh, that is sad. The Homepage introduced that this version is also for dummies :-( |
| 20:42.43 | WIMPy | oliver1/WiretapWork: I don't know what CentOS does, but unless it loads basic networking support as a module, the kernel crashed. |
| 20:43.01 | WiretapWork | oliver1, for the most part that is true, but theyre referring to asterisk for dummies, not linux for dummies |
| 20:43.10 | WIMPy | Which means it has some issue with the hardware. |
| 20:43.20 | WiretapWork | a basic knowledge of the boot sequence is always handy when troubleshooting boot issues |
| 20:43.26 | WiretapWork | yep, I agree WIMPy |
| 20:43.30 | oliver1 | WIMPy: it ist CentOS..., myby the problem is, that I configured the Virtual Box for Debian Linux |
| 20:43.41 | WiretapWork | oh..... its a virtual machine |
| 20:43.46 | oliver1 | yes |
| 20:43.48 | WiretapWork | virtual machines are not supported |
| 20:43.50 | oliver1 | for testing |
| 20:43.52 | WiretapWork | they may work |
| 20:44.02 | WiretapWork | but you should set it up on a physical box for performance's sake anyway |
| 20:45.09 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 20:45.14 | wesphillips | does anyone know a url to documentation for the options passed to the sip channel in a dial command? I can find the docs for the options at the END of the dial command, but not for the options passed to the channel IN the dial command. for example, in the command Dial(SIP/${EXTEN}/n,,K), I can find the docs for the Dial options (the "K" in the example), but not the options to the SIP channel (the "n" in the command). |
| 20:45.23 | seraphie | b0ot: from what I can tell, the CCNP-Voice and CCNA-Voice have nothing to do with asterisk. they are *network* certifications. |
| 20:45.23 | seraphie | b0ot: http://www.digium.com/en/training/certifications/dcap.php |
| 20:45.31 | oliver1 | ok, last week I tried on a other maschine on VB and it works..., so anyway. I have to learn many things... |
| 20:45.58 | oliver1 | I will try it on a physical machine |
| 20:46.04 | b0ot | lol serafie I know the CCNP-Voice and CCNA-Voice have nothing to do with asterisk... they are the cisco certified associate and professional degrees |
| 20:46.08 | oliver1 | thanks a lot for help |
| 20:46.10 | b0ot | both are highly respected in the field |
| 20:46.20 | WiretapWork | oliver1, things are generally a lot less painful on a physical bosx |
| 20:46.20 | b0ot | but I haven't ever heard of anyone mentioning dCAP |
| 20:46.23 | b0ot | or dCAA |
| 20:46.43 | WiretapWork | b0ot, any letters on your CV will cause questions to be asked by recruiters |
| 20:46.51 | WiretapWork | even CCNA or CCNP |
| 20:46.56 | oliver1 | WiretapWork: thank u. I will triy it with a real maschine |
| 20:46.57 | WiretapWork | the fact of the matter |
| 20:47.12 | seraphie | b0ot: you asked "are there any respected asterisk certifications?" |
| 20:47.14 | WiretapWork | you shouldn't get qualifications purely for the wank-factor, you should get them for what you learn |
| 20:47.36 | b0ot | true... but you don't always get paid on what you know |
| 20:47.41 | b0ot | you get paid on what people think you know |
| 20:47.49 | WiretapWork | neither |
| 20:47.50 | b0ot | or think you can do |
| 20:47.54 | WiretapWork | you get paid on what you can show you know |
| 20:48.02 | b0ot | no |
| 20:48.15 | b0ot | anyway nvm this topic is off channel |
| 20:48.17 | WIMPy | WiretapWork: Where is it that way? |
| 20:48.27 | schmidts | WiretapWork if you get paid for a project you are right, but if you get paid for a month of work as an employee this is not 100% true ;) |
| 20:48.39 | citywok | WIMPy: everywhere that will not throw your resume in the circular file for not having a certification, and you make it to the interview |
| 20:48.41 | seraphie | wesphillips: https://wiki.asterisk.org/wiki/display/AST/Application_Dial |
| 20:48.46 | WiretapWork | WIMPy, every job interview I've ever done has been a carefully constructed psychological experiment |
| 20:49.11 | WiretapWork | I would never work anywhere that uses the circular iling cabinet for people without quals |
| 20:49.21 | b0ot | which in the end you form an opinion of what the person you are interviewing knows and is capable of |
| 20:49.26 | citywok | if you make it to the interviews they should be able to figure out whether you know it or you just passed the cert test :P |
| 20:49.28 | WiretapWork | those places tend to have an attitude about them that doesn't gel well with me |
| 20:49.30 | b0ot | which may or may not align with their skillset |
| 20:49.34 | WIMPy | WiretapWork: But I guess you're not working for an outsources HR, AKA headhunter. |
| 20:49.44 | oliver1 | WiretapWork: I solve the problem; I activatet IO-APIC. No it runs |
| 20:49.49 | WiretapWork | WIMPy, I hate outsource-HR firms |
| 20:49.54 | WiretapWork | hate them with a vengance |
| 20:50.09 | b0ot | Letters can you get interviews, interviews get you jobs |
| 20:50.20 | WiretapWork | b0ot, letters never got me an interview |
| 20:50.31 | WiretapWork | aside from a phone interview or 20 with an outsource HR firm |
| 20:50.42 | b0ot | what letters |
| 20:50.47 | WiretapWork | CCN* |
| 20:51.08 | b0ot | CCN* only comes into play after you get engineering or like bachleors degree |
| 20:51.09 | schmidts | CCNX |
| 20:51.09 | citywok | i'm pretty sure that if a hiring manager is looking for a person that knows asterisk, they will interview you simply for knowing it, and won't care (or necessarily even know what dcap is) |
| 20:51.23 | WiretapWork | b0ot, OMFG ARE YOU SERIOUS? you actually believe that? |
| 20:51.34 | b0ot | however engineering degree + CCNX will get you lots of interviews |
| 20:51.47 | WiretapWork | lots of interviews maybe, not so many jobs |
| 20:51.48 | b0ot | I didn't say I think it was right or fair... but yes I would agree with that |
| 20:52.06 | WiretapWork | I have a 100% hitrate of getting the job if I get a face to face with the hiring co. |
| 20:52.18 | WiretapWork | until recently I didn't even have the CCN* next to my name |
| 20:52.25 | WiretapWork | and it had no net effect on my hirability |
| 20:52.30 | b0ot | lol |
| 20:52.30 | keith4 | not that this conversation isn't *fascinating*, but.... |
| 20:52.50 | keith4 | who cares? |
| 20:52.51 | b0ot | anyway as i said this is offtopic |
| 20:53.08 | WiretapWork | keith4, well, I guess b0ot cares :P |
| 20:53.15 | citywok | i generally go with the theory that certs don't matter much :) |
| 20:53.25 | b0ot | most certs don't |
| 20:53.25 | WiretapWork | citywok, prettymuch |
| 20:53.27 | leifmadsen | I don't have a single cert... and look at me!! |
| 20:53.29 | citywok | either you know it in the interview or you don't. not knowing it well and having the cert isn't going to do you much. |
| 20:53.30 | leifmadsen | :) |
| 20:53.39 | leifmadsen | citywok: +1 |
| 20:53.40 | citywok | leifmadsen: a no talent ass clown canadian?!? yup, that's proof of something :D |
| 20:53.40 | b0ot | true statement citywok |
| 20:53.41 | WiretapWork | hahaha, indeed leifmadsen |
| 20:53.46 | leifmadsen | citywok: heck ya |
| 20:53.53 | citywok | lol |
| 20:54.01 | citywok | :heart: |
| 20:54.05 | wesphillips | seraphie: That link gives me the Dial command options, but not the chan_sip options(it tells me about the "K" option, not the "n" option in my example) |
| 20:55.42 | citywok | i love the saying no talent ass clown, it's so perfect. |
| 20:55.42 | keith4 | wesphillips: what's your *real* question? |
| 20:56.48 | wesphillips | keith4: I woul like to find a reference to the options passed to chan_sip in the dial command. I can find the Dial command options(they come AFTER the timeout value), but not the chan_sip options (they come BEFORE the timeout value) |
| 20:58.13 | wesphillips | for example, tom implement a jitterbuffer in sip, you have to pass the dial command a "j" like this: Dial(SIP.${EXTEN}/j,*timeout val*,*dialplan opts*) |
| 20:58.58 | seraphie | wesphillips: are you referring to the dialstring? |
| 20:58.58 | seraphie | SIP/johnsphone, et al/ |
| 20:59.20 | wesphillips | correct |
| 20:59.41 | wesphillips | Dial(SIP/${EXTEN}/j,*timeout val*,*dialplan opts*) |
| 20:59.48 | wesphillips | helps if i don't fat finger.... |
| 20:59.53 | WIMPy | Do you mean sip/peer/something or sip/peer/exten/something? |
| 20:59.54 | keith4 | I've only ever seen /n or /nj for Local channels |
| 21:00.23 | keith4 | n = "no release" |
| 21:00.26 | WIMPy | I've seen others, but I didn't know sip had any such options. |
| 21:01.00 | keith4 | e.g., http://www.voip-info.org/wiki/view/Asterisk+local+channels |
| 21:01.07 | seraphie | wesphillips: here is a little documentation: http://www.russellbryant.net/blog/2007/10/09/asterisk-jitterbuffer-support-for-applications/ |
| 21:01.11 | citywok | i use /n to save channel variables IIRC, but it's been a while |
| 21:01.18 | seraphie | (Google is helpful) |
| 21:01.54 | wesphillips | i have been testing sip-to-sip jitterbuffers, and I cna only get asterisk to create a buffer if I pass it the "j" option. What I want is to be able to have the dialcommand create a buffer for the receive rtp stream, not the transmit stream in the dial command. |
| 21:02.19 | SunTsu | I currently call multiple phones by using SIP/phone1&SIP/phone2&... my problem is: when I take the call on one of the phones all others show "call missed" - is there a way around it? |
| 21:02.45 | WIMPy | SunTsu: Upgrade |
| 21:02.53 | wesphillips | and I was hoping that a lowercase "j" would do TX and an uppercase "J" would to RX, but i would like to see documentation supporting this before I waste my time trying to test it. |
| 21:02.57 | SunTsu | WIMPy: to 1.8? |
| 21:02.57 | WIMPy | That has been fixed a long time ago. |
| 21:03.09 | citywok | WIMPy: orly? |
| 21:03.24 | WIMPy | SunTsu: It was fixed befor 1.8. |
| 21:03.29 | citywok | WIMPy: with Lync when i do sip/1593&sip/1593@lync Lync sees it as a mixed call |
| 21:03.34 | seraphie | wesphillips: more: http://www.asterisk.org/astdocs/node176.html |
| 21:03.37 | citywok | missed* |
| 21:04.09 | WIMPy | citywok: Maybe it doesn't understand what Asterisk tells it. |
| 21:04.11 | seraphie | these may be specific to misdn, though. |
| 21:04.35 | SunTsu | WIMPy: OK, thanks, I'll upgrade |
| 21:04.43 | citywok | hmm. I'll have to look at a packetcapture and see, if it is a lync issue i'll post it to microsoft to fix |
| 21:04.44 | puzzled | http://www.chromis.com/blog/?p=2540 |
| 21:05.00 | citywok | do you know when it was added roughly? i'm on 1.6.2.11 still |
| 21:05.41 | WIMPy | No, I used to apply a 3rd party patch befor it got to the official tree. |
| 21:06.05 | WIMPy | But 1.6.11 might be before. |
| 21:06.36 | wesphillips | thanks for the help. It looks like the "n" and "j" are the only options as far as I can tell. |
| 21:06.43 | citywok | ah, okay. I don't want to upgrade since i had to write a patch for a meetme feature i wanted and it isn't committed to the releases |
| 21:07.16 | citywok | yet anyways, it will be at some point. i just don't want to have to re-write that patch for each upgrade :P |
| 21:07.26 | p3nguin | citywok: I think interviewers should test on other skills besides those of a technical nature. If I were interviewing, I'd have my candidate copy down a few sentence that I speak... I'd give sentences that tested for proper usage of your/you're, there/they're/their, its/it's, and several other bits of 5th-grade English grammar. If they can't pass a 5th grade English test, they don't need to be working for me. |
| 21:07.32 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:08.09 | citywok | p3nguin: dear god those are some of my biggest pet peeves. i know a lot of people that can't use those properly. including my boss. the other ones that annoy me are "should of" instead of should have... |
| 21:08.10 | WIMPy | citywok: I submitted a patch to that feature for other channels 2009-10-29. |
| 21:08.11 | p3nguin | If they can pass it, then we'll talk about higher level stuff. |
| 21:08.50 | citywok | "i should of gone to the grcoery store" -- really? i mean REALLY?!? |
| 21:09.12 | SunTsu | dang, netbsd doesn't have any upgrade in pkgsrc. Still 1.6.2.16.1 |
| 21:09.14 | el3slave | so asterisk wont show call progression over ssh? |
| 21:09.14 | p3nguin | I'd be lenient on punctuation, but wrong words are grounds for dismissal. |
| 21:09.58 | citywok | it's is a common mistake that i can handle |
| 21:10.21 | citywok | but "your not very nice" makes me want to start shooting |
| 21:11.20 | tzanger | p3nguin: that's an interesting thing to check for |
| 21:11.21 | p3nguin | I understand that sometimes fingers overrun brain, so I would review the results of the writings, not give any feedback, then allow the candidate to make corrections. |
| 21:11.24 | citywok | Do you read SA? this thread title pisses me off every time i see it: "Shit that you come across daily that pisses you off" -- it makes me want to post "This thread title pisses me off", but it'd cost me :10bux: |
| 21:11.28 | tzanger | I'm not against it, but I never would have thought of it |
| 21:11.44 | tzanger | although I think that the initial emails back and forth would have been a good indicator as well |
| 21:12.44 | keith4 | why wouldn't that guy just have enabled the jitter buffer stuff in sip.conf? |
| 21:12.47 | tzanger | damn, someone should have told wesphilips that jitter buffers only exist on "digital hop off" points |
| 21:12.53 | tzanger | sip-to-sip will never have a JB in asterisk |
| 21:14.31 | WIMPy | citywok: Just had a look while on the phone. It loks like I didn't keep the patches. |
| 21:15.46 | el3slave | can anyone tell me why im not seeing the call progression in either terminal or over ssh in * cli? |
| 21:15.55 | el3slave | verbosity is set at 9999 |
| 21:16.05 | citywok | el3slave: what's your logger.conf set to? |
| 21:18.26 | el3slave | citywok dont have one ;) |
| 21:18.32 | el3slave | guess ill have to set that up |
| 21:18.35 | citywok | el3slave: that could be a problem lol |
| 21:18.41 | el3slave | hah |
| 21:18.43 | el3slave | thanks |
| 21:19.54 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
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| 21:21.42 | el3slave | wee |
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| 21:30.59 | WiretapWork | citywok, just read about your pet peeve, I have the same one and it helped get me this job :P |
| 21:31.25 | citywok | which one? that or of/have? |
| 21:31.33 | WiretapWork | apostrophe abuse |
| 21:31.45 | WiretapWork | the guy asks me "so what are a couple of your pet peeves" |
| 21:31.54 | WiretapWork | me: "with regards to work, or in general?" |
| 21:31.59 | Chainsaw | WiretapWork: Hey now, spurious apostrophe syndrome is a serious disease. |
| 21:32.04 | WiretapWork | him: "life, the universe, politics, anything" |
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| 21:32.13 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 21:32.14 | WiretapWork | me: "well..... bad drivers and apostrophe abuse" |
| 21:32.25 | WiretapWork | *cue all three of them cracking up laughing* |
| 21:33.38 | citywok | lol |
| 21:33.55 | citywok | i like to ask people if it is possesive or plural, it's funny when they go o.O? what are you talking about |
| 21:34.16 | Chainsaw | possessive |
| 21:34.38 | WiretapWork | citywok, best part was that the managing director didn't know what an apostrophe was and thought I was talking about commas :P |
| 21:35.21 | citywok | lol |
| 21:45.33 | ChannelZ | I've always enjoyed the forward slash/back slash debate |
| 21:46.22 | citywok | haha, yea i call \ back |
| 21:46.28 | citywok | it looks like it's leaning back to me |
| 21:46.41 | ChannelZ | Indeed |
| 21:47.07 | keith4 | whether or not you call it that... that's what it is ;-) |
| 21:47.17 | WIMPy | Nah, both forward and backward look like | |
| 21:47.49 | ChannelZ | counter-clockwise pipe and clockwise pipe.. yeah, that's the ticket! |
| 21:48.12 | el3slave | hah |
| 21:48.38 | citywok | i love how inviting a couple people over for dinner ends up being 7 people, lol |
| 21:49.31 | ChannelZ | I don't have that many chairs |
| 21:49.41 | citywok | me either |
| 21:50.14 | citywok | oh well, couch + 4 dinner chairs & my desk chair. lol. although i think i've had 15 people over playing kinect and drinking, but we had to re-arrange my living room for that. |
| 21:52.50 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 21:55.05 | el3slave | if they wanna eat, they gotta bring chairs, or STAND!@ |
| 21:57.25 | ChannelZ | I guess that's why god made chicken nuggets |
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| 22:37.21 | *** join/#asterisk pdtpatrick___ (~pdtpatric@mainstwan.farheap.com) |
| 22:37.35 | pdtpatrick___ | Question for yo usmart people |
| 22:38.01 | pdtpatrick___ | if i made an edit to extensions.conf -- how do i reload so asterisk picks it up? i did dialplan reload on CLI |
| 22:38.03 | pdtpatrick___ | no luck |
| 22:38.08 | pdtpatrick___ | have done reload no luck |
| 22:38.23 | citywok | pdtpatrick___: dialplan reload |
| 22:38.36 | pdtpatrick___ | done that .. it keeps bringing up the old context |
| 22:38.43 | pdtpatrick___ | even after i've renamed it |
| 22:39.07 | ChannelZ | is this asterisknow/freepbx? |
| 22:39.35 | pdtpatrick___ | no just asterisk |
| 22:40.21 | ChannelZ | not realtime I assume |
| 22:41.23 | pdtpatrick___ | what do u mean ? |
| 22:41.58 | ChannelZ | Double check your syntax of everything (and/or pastebin your extensions.conf if you want us to look), maybe do a 'dialplan show' t make sure you don't have something unpexpected happening from AEL or something else |
| 22:41.59 | pdtpatrick___ | im on 1.6.2 |
| 22:42.28 | ChannelZ | realtime = parts of the config come from a database rather than config files |
| 22:43.12 | ChannelZ | If you don't know you're using it then you're almost certainly not (that was related to the AsteriskNOW/freepbx question) |
| 22:43.41 | pdtpatrick___ | im sure it is not freepbx |
| 22:43.45 | pdtpatrick___ | or asterisk now |
| 22:43.54 | pdtpatrick___ | I changed the config manually |
| 22:43.57 | pdtpatrick___ | do i need to do something to db ? |
| 22:45.27 | ChannelZ | no unless you've specifically setup realtime, there is no db |
| 22:46.17 | pdtpatrick___ | there's postgres on here |
| 22:46.22 | pdtpatrick___ | is there something i would have to do ? |
| 22:46.30 | ChannelZ | again, no |
| 22:46.56 | pdtpatrick___ | okay so if it seeing the old context even after dialplan reload ... what should i do then? |
| 22:47.09 | ChannelZ | check your syntax |
| 22:47.39 | ChannelZ | look for a warning when you 'dialplan reload' |
| 22:47.41 | *** join/#asterisk |Physis| (~|Physis|@201009154240.user.veloxzone.com.br) |
| 22:47.54 | ChannelZ | You said you renamed an entire context... from what to what? |
| 22:48.44 | |Physis| | I'm having trouble using the voicemail recording the data in the database postgresql using odbc, selecting MENUSELECT_OPTS_app_voicemail = ODBC_STORAGE. Displays the following error when I leave a message on voicemail: app_voicemail.c: 3661 insert_data_cb: Direct SQL Execute failed! ??? |
| 22:48.58 | |Physis| | <PROTECTED> |
| 22:49.11 | |Physis| | <PROTECTED> |
| 22:49.20 | |Physis| | :( |
| 22:49.24 | pdtpatrick___ | it was jnt_800_dial ... but i changed it to jnt .. now it only sees jnt_800_dial .. so when i dial the number it keeps saying rejected because context not found |
| 22:49.55 | ChannelZ | well that's a different problem then you're asking |
| 22:50.22 | pdtpatrick___ | please explain .. very grateful by the way |
| 22:50.24 | ChannelZ | You renamed a context of extensions but your device is still going into the old context (which now doesn't exist) |
| 22:50.33 | pdtpatrick___ | right |
| 22:50.47 | pdtpatrick___ | so how do i force it to read the new context |
| 22:50.49 | ChannelZ | so it seems like you're getting what you should |
| 22:50.59 | ChannelZ | You need to edit the device and put it in the right context |
| 22:51.12 | ChannelZ | sip.conf if you're dealing with a SIP device, etc. |
| 22:51.51 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 22:52.13 | pdtpatrick___ | we already edited sip.conf .. and made the necessary changes. To match in sip.conf and extensions.cnf |
| 22:52.20 | pdtpatrick___ | however it just keeps looking for the old name |
| 22:52.55 | ChannelZ | then reload your sip devices... sip reload |
| 22:54.38 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 22:55.00 | |Physis| | any help ? |
| 22:55.06 | pdtpatrick___ | have done that .. sip reload , dialplan reload |
| 22:55.09 | pdtpatrick___ | same stuff |
| 22:55.11 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 22:55.34 | ChannelZ | then back to checking your syntax and for typeos |
| 22:55.53 | ChannelZ | Or pastebin your configs if you want us to give them a glance |
| 22:55.53 | ChannelZ | ~pb |
| 22:55.54 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 22:56.49 | pdtpatrick___ | weird is it works on one server but not the other.. 1.4 but on 1.6 it doesn't work |
| 22:57.20 | ChannelZ | Are you trying to connect two systems together? |
| 22:57.47 | pdtpatrick___ | yes |
| 22:58.19 | Kobaz | http://i6.aijaa.com/b/00097/8091042.png |
| 22:58.20 | ChannelZ | so you've got one of the peers setup wrong but we can't guess without seeing anything |
| 22:58.42 | pdtpatrick___ | should it be peer or friend btw |
| 22:58.42 | ChannelZ | heh |
| 22:59.13 | ChannelZ | oh no, can of worms! |
| 22:59.21 | citywok | lol |
| 23:00.38 | pdtpatrick___ | right now i have it like _800blah, n,Dial(SIP/4213) |
| 23:01.15 | pdtpatrick___ | however if i do something like Dial(Local/600@context) |
| 23:01.17 | pdtpatrick___ | this works |
| 23:02.05 | pdtpatrick___ | but i have a context on the box named onp_main ... and i would love to have it like Dial(SIP/onp_main) or does it have to be a Goto ? |
| 23:02.12 | ChannelZ | Does the SIP device 4213 exist? What context is that extension in? What context does the device dialing 800blah belong to? There could be 10 different problems |
| 23:02.31 | pdtpatrick___ | 4213 is under onp_main |
| 23:02.40 | ChannelZ | what does that mean |
| 23:02.48 | ChannelZ | SIP/4213 is a device name, not an extension |
| 23:03.18 | pdtpatrick___ | ahhh .. how would i put an extension then? |
| 23:03.20 | ChannelZ | (rather 4213 is a SIP device name to be clearer) |
| 23:03.30 | pdtpatrick___ | onp_main is the context |
| 23:03.32 | ChannelZ | _800blah is the extension |
| 23:03.48 | pdtpatrick___ | so under here .. it says if 4213 is called, then it has all these steps like answer, play this music etc |
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| 23:04.11 | ChannelZ | If you want to dial 1000 to have the 4213 phone ring, you would exten => 1000,1,Dial(SIP/4213) |
| 23:04.43 | ChannelZ | Extensions aren't devices and are totally unrelated |
| 23:05.39 | pdtpatrick___ | we have n instead of the 1 |
| 23:05.43 | pdtpatrick___ | big deal? |
| 23:05.50 | ChannelZ | We're talking in generalities here, we can't fix your dialplan or config because we have no idea what your config even looks like |
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| 23:06.11 | ChannelZ | Not if something happens before it |
| 23:06.12 | pdtpatrick___ | also 4213 is defined on another box.. how can i make it connect and see context from another box? |
| 23:07.17 | ChannelZ | You need to start by only dealing with one system and get a simple setup working, understand devices and extensions and contexts before you try networking systems |
| 23:08.38 | ChannelZ | This isn't meant as rude or condescending |
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| 23:09.05 | pdtpatrick___ | have dundi installed on the boxes |
| 23:09.12 | pdtpatrick___ | ChannelZ no worries |
| 23:09.16 | pdtpatrick___ | im learning as I go |
| 23:09.31 | ChannelZ | Sure but I think you're starting too big |
| 23:09.34 | pdtpatrick___ | i just would like to get this working to get back to my reading... troubleshooting is one of the best ways to learn |
| 23:09.42 | pdtpatrick___ | possible :) |
| 23:09.59 | ChannelZ | if you don't have a handle on how the basics work it's hard to talk about something complicated like linking two different systems |
| 23:12.13 | *** join/#asterisk SaschaL (~Sascha@5acd5c71.bb.sky.com) |
| 23:12.35 | SaschaL | Hi all, huge problem with an asterisk installation I have running with freepbx and elastix running on top |
| 23:13.09 | SaschaL | I've been hacked by someone and they've made premium rate calls. Luckily I just noticed it and the calls have only been going on for an hour, so I caught it before too much damage was done |
| 23:13.09 | pdtpatrick___ | okay simple questio nthen .. if i have dundi -- can one box use context from another.. for instance if i defined onp_main on extensions.conf on box2 .. can box1 just point to onp_main ?? |
| 23:13.18 | SaschaL | I've just deleted my trunks so they can't make calls |
| 23:13.44 | SaschaL | but I don't understand how they've gotten the extention details, as I patched up the elastix 1.6 vunerability, and changed all the secrets |
| 23:13.50 | SaschaL | Any idea's how I can find out? |
| 23:14.56 | WiretapWork | SaschaL, brute force or DISA probably |
| 23:15.15 | citywok | SaschaL: were you using 2 to 4 digit extensions with fairly simple passwords? |
| 23:15.28 | SaschaL | WiretapWork: What's DISA? and as far as brute force, I'll be setting highly complex passwords in future |
| 23:15.44 | WiretapWork | you should make sure only extensions which absolutely need to be connected from the outside have outside addresses allowed, all else should be denied |
| 23:15.47 | SaschaL | extentions were 4 digits, passwords were 1stCall, which I guess was too easy |
| 23:15.57 | citywok | SaschaL: i'd also suggest using fail2ban to help protect yourself from scanning. and if you don't need peers connecting from the outside prevent it :P |
| 23:16.22 | citywok | i use 4 digit extensions, but our passwords are like 40 characters long and different by extension |
| 23:16.25 | WiretapWork | SaschaL, dictionary words, really? |
| 23:16.28 | SaschaL | citywok: Unfortunately this is an offsite PBX, so it does need peers, and the clients IP is dynamic not static |
| 23:16.29 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 23:16.37 | citywok | and i use allow/deny to restrict it |
| 23:16.55 | citywok | SaschaL: if you know the ISP uses a fairly stagnant IP block (say 74.0.0.0) limit it to that at least |
| 23:17.00 | SaschaL | WiretapWork: A large mis-sight on my part |
| 23:17.21 | citywok | s/stagnant/consistent/ |
| 23:17.30 | WiretapWork | I love the 'deleted my trunks so they cant make calls' response |
| 23:17.46 | WiretapWork | disabling the trunks would have worked just fine XD |
| 23:17.56 | SaschaL | WiretapWork: It was just a quick thing to do until I get the system all patched up |
| 23:18.17 | SaschaL | I'll change all the extention passwords in a minute and reboot the server |
| 23:18.35 | SaschaL | what is DISA though? |
| 23:18.46 | SaschaL | Apologies for my naivity |
| 23:19.26 | WiretapWork | Direct Inwards System Access |
| 23:19.43 | WiretapWork | allows someone to make an inbound call and then from there make an outbound call as if it were internalk |
| 23:19.47 | WiretapWork | -k |
| 23:19.57 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 23:20.39 | SaschaL | WiretapWork: it wouldn't have been someone who has access already. So it must have been bruteforce |
| 23:21.00 | WiretapWork | SaschaL, of course, but even having DISA on is a big risk |
| 23:21.16 | WiretapWork | oldschool PBX are just as vulnerable to DISA exploits |
| 23:21.54 | SaschaL | I don't believe it's on |
| 23:22.06 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 23:22.11 | SaschaL | no, in fact, it's not |
| 23:22.28 | ChannelZ | wandered off, catching up - do you have anonymous sip turned on? (guest access, unathenticated essentially) |
| 23:24.18 | WiretapWork | ChannelZ, with freepbx guest mode will be on, but it'll be blocked by freepbx if freepbx's own anonymous sip access is off |
| 23:24.35 | WiretapWork | which is handy for handling ISN calls, etc, while still blocking most anonymous calls |
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| 23:25.16 | WiretapWork | I have it set up so if anonymous calls match my ISN inbound patterns theyre permitted, otherwise theyre denied if anonymous |
| 23:25.25 | ChannelZ | I guess the main question was were they really authing as one of your own SIP peers or were the calls being made another way.. like bad dialplan separation, but I would hope/assume fpbx deals with that for you |
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| 23:40.24 | SaschaL | WiretapWork: Ok, just changed all the secret's |
| 23:40.36 | SaschaL | WiretapWork: Just out of interest, is there a way to block international calls |
| 23:40.43 | SaschaL | and calls to certain prefix' |
| 23:41.10 | WiretapWork | SaschaL, you could tear off the country code and international prefix |
| 23:41.25 | pdtpatrick__ | what does ${CALLERID(dnid)} mean? .. actually im trying to figure out where dnid is what it does |
| 23:41.56 | SaschaL | WiretapWork: The SIP provider I'm using doesn't require country code for UK |
| 23:41.57 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 23:42.02 | SaschaL | so it has to be 11 digits |
| 23:42.13 | WiretapWork | SaschaL, ehh? |
| 23:42.20 | SaschaL | could I technically disable all calls that don't start with 01, 02, 03 and 07 |
| 23:42.26 | WiretapWork | pdtpatrick__, it is well doc'd |
| 23:42.44 | WiretapWork | SaschaL, yep, you could do that in the trunk's pattern matching, so that it only accepts calls that match that |
| 23:42.45 | citywok | SaschaL: in asterisk you have to ENABLE a pattern |
| 23:42.54 | citywok | SaschaL: so simply by not enabling it, you are disabling it |
| 23:42.56 | SaschaL | WiretapWork, they don't use enum. So the format isn't 441234567891 it's 01234567891 |
| 23:43.14 | WiretapWork | SaschaL, you mean E164, not Enum |
| 23:43.21 | SaschaL | yes sorry |
| 23:43.47 | WiretapWork | SaschaL, so? you could still have a dialpattern match that strips '00xx' from the start of any dialled number, forcing them local |
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