| 00:02.07 | rogersja | has anyone had any luck setting CID on an outgoing google voice call from asterisk? |
| 00:07.43 | *** join/#asterisk pLr (~acer@unaffiliated/plr) |
| 00:07.50 | *** part/#asterisk pLr (~acer@unaffiliated/plr) |
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| 00:10.28 | luckyaba | for what ever reason exten => 123,1,MeetMe(,Mde) won't create a room. I keep getting "That is not a valid conference number" |
| 00:10.33 | luckyaba | what am I missing |
| 00:10.46 | luckyaba | should that not auto create a conference room and join you to it? |
| 00:14.05 | luckyaba | we should have IRC up on the wall when nothing is going on |
| 00:14.17 | luckyaba | so you bastards know when I am trying to talk to you |
| 00:17.27 | Freeaqingme | luckyaba, most clients support highlighting ;) |
| 00:20.28 | luckyaba | lies |
| 00:20.51 | luckyaba | they should support a loud audible announce that says "LUCKY IS TALKING" |
| 00:22.43 | luckyaba | that was actually meant for another channel |
| 00:22.48 | luckyaba | but still funny |
| 00:22.49 | luckyaba | :P |
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| 00:28.38 | whytek | Quick question for anybody - How can I get sip brute force attempts if my asterisk is running on a virtualbox behind a NAT with no ports forwarded? |
| 00:29.01 | whytek | NOTICE[2493]: chan_sip.c:15236 handle_request_register: Registration from '"commrades"<sip:commrades@192.168.1.71>' failed for '124.162.56.218' - No matching peer found |
| 00:29.14 | Freeaqingme | most logical would be that you have a box in your network filled with trojans whytek |
| 00:29.40 | whytek | hi thanks, but how so the 124.162.56.218 ip address there? |
| 00:29.58 | Freeaqingme | perhaps your nat isn't as solid as you'd hoped for |
| 00:31.45 | jizzzum6 | connect a machine outside your NAT and try to SIP register |
| 00:32.03 | whytek | It's just a test box.. i'm new to asterisk, but i know my networks, is it correct for this to happen, that the public ip has to be forwarding requests on port 5600 to my internal vbox? |
| 00:32.38 | jizzzum6 | should be 5060 udp or tcp for SIP messages |
| 00:32.53 | jizzzum6 | then RTP would be needed for actual communications (beyond call setup stuff) |
| 00:33.28 | whytek | udp |
| 00:33.30 | pabelanger | jizzzum6: depends on how you configure asterisk, but usually UDP |
| 00:33.54 | whytek | sip is udp, right? |
| 00:34.12 | pabelanger | no, it can be TCP and TLS too |
| 00:34.19 | Marvelous | hello |
| 00:34.32 | Marvelous | i need free trunk to call toll free |
| 00:34.39 | whytek | ok, let's see we are listening on 2000 and 5038 tcp and a bunch or ports udp |
| 00:34.53 | jizzzum6 | maybe someone is tricking your UDP NAT into talking to them |
| 00:35.09 | whytek | wierd that they would even find it! |
| 00:35.16 | jizzzum6 | UDP is harder to control direction (due to lack of a clear SYN like in TCP) |
| 00:35.32 | pabelanger | whytek: 2000 is skinny and 5038 is manager |
| 00:35.52 | whytek | thanks |
| 00:36.16 | jizzzum6 | Marvelous: http://www.tollfreetollfree.com/ |
| 00:36.34 | jizzzum6 | free toll free nums in US via SIP |
| 00:39.04 | whytek | oh well, i'm actually not on my home network, behind a DSL 2wire modem in an office ,and i can't actually see the pf setup right now, but last time i looked is was default, as in empty., if it happens again i'm look into it some more. |
| 00:45.03 | Marvelous | thanks |
| 00:45.22 | Marvelous | thanks jizzzum6 |
| 00:47.24 | adr3nalin3 | I am having trouble with Aastra 6730i phones, in asterisk an outgoing call is still going on however the Aastra phone says call failed after about 1-2 seconds |
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| 00:59.21 | Marvelous | jizzum6 what else good i can use it :D |
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| 01:14.05 | cj | can someone tell me how to test my FXS port? I think I might have damaged it by plugging it into the incoming FXO wire |
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| 02:07.16 | Marvelous | hey |
| 02:07.54 | kaldemar | cj: plug a phone in it. |
| 02:09.38 | kaldemar | cj: or plug it into an fxo interface. |
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| 03:27.24 | golikwid|mac | anyone in here familiar with the ss7 lib? |
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| 04:01.33 | c2tarun | I was referring to this page to establish a PBX by asterisk, and when I executed stop now command I am getting an error that there is no such command. |
| 04:17.43 | ChannelZ | core stop now |
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| 04:25.00 | dan__t | Hi. |
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| 04:48.47 | rogersja | c2tarun, if you want to use stop now, you'd have to employ a cli alias |
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| 05:46.32 | trijezdci | hi, does anyone know of a Cisco WebEx Connect client integration with SIP or other open standard protocols (without the use of Cisco's Call Manager) ? |
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| 06:32.45 | Exten | is there a one liner/non cli to get the "show channels concise" ? |
| 06:33.14 | Wiretap7 | Exten, `asterisk -rx "show channels concise"` what you after? |
| 06:34.05 | Sertys | i'm kinda wondering |
| 06:34.22 | Sertys | i enable extconfig sippeers source from a mysql db |
| 06:34.33 | Sertys | if for some reason the mysql server is unavailable |
| 06:34.46 | Sertys | the whole asterisk dies |
| 06:34.59 | Sertys | even though i have plenty of peers defined in the sip.conf |
| 06:35.37 | Sertys | is there a way to set a priority for the extconfig or would it help if i enabled caching |
| 06:36.48 | Exten | <Wiretap7> i wanna have only the show channels concise output - and maybe put it in a cgi or >> to a text file so i would be able to see a "monitor" for the calls |
| 06:37.00 | Wiretap7 | Exten, try that command |
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| 06:37.28 | Exten | the problem with that command is that if i have many many calls - it sometimes puts a part of whats going on with the cli |
| 06:37.33 | kaldemar | Exten: if you will use a cgi, use the same command via AMI. |
| 06:37.36 | Exten | (other calls and stuff) |
| 06:37.51 | Exten | <kaldemar> havent had a chance to work with AMI |
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| 06:41.29 | kaldemar | Exten: now you do. |
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| 06:47.04 | Exten | lol |
| 06:58.57 | Wiretap7 | Exten, I think you're completely missing the part where the entire command from the linux shell is |
| 06:59.04 | Wiretap7 | asterisk -rx "show channels concise" |
| 06:59.17 | Wiretap7 | it will do nothing other than print the concise channels list to stdout |
| 07:01.17 | JerJer | and perhaps a little extra too :) |
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| 07:07.19 | jacc0 | hi all :) good morning |
| 07:07.31 | jacc0 | is the SVN up already? |
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| 07:12.11 | Exten | <Wiretap7> it sometimes gives a few more lines with what happens on cli |
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| 07:13.25 | Wiretap7 | would that not be a bug? |
| 07:14.34 | Exten | good question |
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| 07:51.03 | tuxx- | hey guys, does anyone know if there is more documentation about call-limit then specified on voip-info.org? I'm trying to find out if there is a call-limit for incoming and outgoing calls, because we want to make a difference between those 2. Anyone maybe has a link to some more documentation? Or can maybe tell me that our idea is stupid ;-) |
| 07:52.45 | tuxx- | by call-limit, i mean the option for a sip peer. |
| 07:53.03 | tuxx- | call-limit = number : Number of simultaneous calls through this user/peer. |
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| 07:54.34 | Exten | i think you can count the number of calls on a specific context with a group variable i forgot exactly the name of it ... |
| 07:55.11 | Exten | its group-something ... |
| 07:55.27 | tuxx- | tnx |
| 07:55.30 | tuxx- | im gonna look for it :) |
| 07:55.40 | Exten | http://www.voip-info.org/wiki/view/Asterisk+groups |
| 07:55.44 | Exten | ok its called group-count |
| 07:56.14 | tuxx- | gonna try it out, tnx for the help :) |
| 07:56.17 | Exten | you can count the number of group-count if it is < or > from something |
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| 08:11.16 | Wiretap7 | interesting |
| 08:11.30 | Wiretap7 | restarting asterisk makes my 7970 regrab its configuration |
| 08:11.35 | Wiretap7 | but I can't do that normally O-o |
| 08:22.41 | trijezdci | hi, does anyone know of a Cisco WebEx Connect client integration with SIP or other open standard protocols (without the use of Cisco's Call Manager) ? |
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| 08:37.39 | jkroon | when configuring dahdi/system.conf - specifically for E1 - how crucial is it to set the timing settings correctly? ie, timing pref - if I always set them to 1 thru 4 is that OK? I've done a cable between two such ports and it's reporting OK but I want to know whether I'm going to run into problems down the line? |
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| 08:49.35 | jacc0 | ~backtrace |
| 08:49.35 | infobot | backtrace is a debugging tool that is useful when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read doc/backtrace.txt). See also https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
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| 09:16.43 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:16.58 | Chainsaw | Morning. Issue tracker unreachable for anyone else please? |
| 09:17.34 | Chainsaw | Looks to be the IPv6 address only. |
| 09:18.08 | Chainsaw | (As in I get ICMP response on 76.164.171.231 but not 2001:470:e0d4::e7) |
| 09:18.08 | jacc0 | nope |
| 09:18.19 | jacc0 | https://just loged in to issues.asterisk.org |
| 09:18.30 | Chainsaw | jacc0: And you are on an IPv6-enabled machine/network? |
| 09:18.43 | jacc0 | I don't think so |
| 09:18.56 | Chainsaw | jacc0: Then it is expected that you would see no difference, indeed. |
| 09:19.03 | jacc0 | justa added valgrind.txt to : https://issues.asterisk.org/view.php?id=19203 |
| 09:19.04 | Chainsaw | pabelanger: ping |
| 09:21.26 | Chainsaw | leifmadsen, Qwell, russellb: Looks like IPv6 is down for issues.asterisk.org; would you mind withdrawing the AAAA for the moment? |
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| 09:56.37 | Exten | is there a Set if , or a way to set if |
| 09:56.50 | Exten | i remebmer there was a way to do this with an IF function somehow |
| 10:01.42 | Chainsaw | ExecIf you mean? |
| 10:08.07 | Exten | like If menu=1 then set( |
| 10:08.10 | Exten | like If menu=1 then set(a=b) |
| 10:08.20 | Exten | mm... execif maybe |
| 10:08.31 | Chainsaw | nods |
| 10:09.00 | Chainsaw | Exten: Here's a working example: |
| 10:09.01 | Chainsaw | exten => s,n,ExecIf($[ "${crmname}" != "UNKNOWN" ]?Set(OUTNAME=${crmname}):Set(NOCRM=1)) |
| 10:10.15 | Exten | Cool 10x !@# |
| 10:10.38 | Chainsaw | Exten: You're welcome. |
| 10:10.42 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 10:11.01 | Chainsaw | Hello Sertys. |
| 10:11.15 | Sertys | hi |
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| 10:37.32 | Chainsaw | leifmadsen: Any chance of issue #19192 blocking 1.8.3.4 so the next release can be deployed here? |
| 10:38.00 | *** part/#asterisk nvucinic (~nvucinic@unaffiliated/nvucinic) |
| 10:48.35 | jacc0 | leifmadsen: and #19203 so we can use 1.8.4 in our production evn. ? :) |
| 10:48.45 | jacc0 | ;) |
| 10:49.04 | jacc0 | @Chainsaw: nice bug report |
| 10:49.08 | Chainsaw | waits 60 seconds for that to load |
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| 10:49.32 | Chainsaw | jacc0: Fellow users on #gentoo-voip found it based on my description of the fault. |
| 10:49.47 | Chainsaw | jacc0: My system refused to dump core; probably because of hardened security patching. |
| 10:51.20 | jacc0 | ;) |
| 10:51.34 | Chainsaw | jacc0: But it manifested in an interested way. I have a few peers that are unreachable on TCP but not UDP. So as the timeout hit, roughly 5 minutes after starting Asterisk, it exploded violently. |
| 10:51.54 | Chainsaw | jacc0: Over and over again. I've had to revert. |
| 10:52.41 | jacc0 | I think th ebug I reported causes the random segfaults i'm experiancing |
| 10:53.41 | Chainsaw | jacc0: Segfaults are a luxury. My SIP stack is dead in the water from time to time. |
| 10:53.46 | jacc0 | but I'm not sure |
| 10:53.50 | Chainsaw | jacc0: I notice because my SIP gateways fail over to the analog phones here. |
| 10:54.11 | Chainsaw | jacc0: So the phone of doom starts ringing: http://www.vroon.org/ringring.jpg |
| 10:54.33 | jkroon | Chainsaw, indeed ... i've got similar IAX/2 issues. |
| 10:54.54 | Chainsaw | is addicted to this slick new Connected Line Update system though |
| 10:54.59 | Chainsaw | So I put up with it for the moment. |
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| 11:30.28 | leifmadsen | Chainsaw: jacc0: I'll take a look, but not sure where we're going to get this week -- don't think we'll have power at Digium again today |
| 11:30.42 | Chainsaw | leifmadsen: Ah, you had a storm? |
| 11:30.50 | leifmadsen | Chainsaw: it would block 1.8.4, not 1.8.3.4 as 1.8.3 is already released |
| 11:31.05 | Chainsaw | leifmadsen: 1.8.4 is what I meant, but not what I said. Sorry, and yes, agreed. |
| 11:31.06 | coppice | then pay your bills :-\ |
| 11:31.12 | leifmadsen | Chainsaw: you haven't seen the news about the worst storm in Alabama in over 30 years? |
| 11:31.20 | Chainsaw | leifmadsen: I don't follow the news, no. |
| 11:31.31 | leifmadsen | most of Alabama has been without power for over a week |
| 11:31.45 | leifmadsen | Digium has been running on generator power since last Wednesday |
| 11:32.01 | leifmadsen | so if you see no issues moving, that's why |
| 11:32.09 | Chainsaw | leifmadsen: *nod* Understood. |
| 11:32.09 | leifmadsen | things won't get back to normal for at least another couple of days |
| 11:33.44 | jacc0 | If they don't : https://ackspace.nl/wiki/File:Geek_whip.jpg |
| 11:33.46 | jacc0 | ;) |
| 11:34.11 | jacc0 | :P |
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| 11:40.25 | leifmadsen | welp mantis isn't working for me so someone will have to remind me about those issues later when it is |
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| 11:41.02 | Amorsen | Is issues.asterisk.org having... issues? |
| 11:41.15 | leifmadsen | <leifmadsen> welp mantis isn't working for me so someone will have to remind me about those issues later when it is |
| 11:41.27 | leifmadsen | Amorsen: yes, probably power issues derailing the internet in Alabama |
| 11:41.32 | jacc0 | ipv6 problem as Chainsaw explained |
| 11:41.48 | Amorsen | Thanks |
| 11:41.54 | Chainsaw | leifmadsen: Can you withdraw the AAAA for the moment? |
| 11:41.57 | jacc0 | I've have no problem accessing it |
| 11:42.04 | leifmadsen | Chainsaw: nope |
| 11:42.16 | Chainsaw | leifmadsen: Okay. |
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| 11:42.25 | Chainsaw | Amorsen: Your page will load after 60 seconds. |
| 11:42.32 | leifmadsen | I don't have access to the DNS records |
| 11:42.48 | Amorsen | No such luck unfortunately Chainsaw |
| 11:43.02 | leifmadsen | Amorsen: probably routing issues -- things in Alabama are still very fragile |
| 11:43.21 | Chainsaw | Amorsen: Depends on what you set your IPv6 timeout to; but if you are dual-stack you should fall back to IPv4 which will work. |
| 11:43.49 | leifmadsen | I'm only IPv4 here |
| 11:43.51 | leifmadsen | still doesn't load |
| 11:47.03 | Amorsen | SYSTEM WARNING: mysql_connect() [function.mysql-connect]: Too many connections |
| 11:47.12 | leifmadsen | sounds like bots then |
| 11:47.17 | leifmadsen | that happens about this time of day |
| 11:47.23 | Amorsen | Review Board is taking a nap |
| 11:47.32 | Chainsaw | Amorsen: Smoke is coming out of the database server. Please allow it time to cool down. |
| 11:47.52 | Amorsen | Is it the same DB server for issues and reviewboard? |
| 11:52.25 | Amorsen | Looks all better now |
| 11:52.33 | oelewapperke | is there a list of "Local/" channels you can dial ? |
| 11:52.51 | oelewapperke | I need some way to park a call, but I don't need announcements and I'm not using a dialplan |
| 11:52.57 | oelewapperke | so "Park" is useless |
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| 11:53.18 | kaldemar | oelewapperke: Local/extension@context, as in your dialplan. |
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| 11:55.13 | oelewapperke | kaldemar: so you don't have a way to just generate a channel with waitmusic or something |
| 11:55.52 | kaldemar | not with chan Local. maybe origination is what you're looking for? |
| 11:59.18 | oelewapperke | kaldemar: no it's not, unless I'm missing something |
| 11:59.32 | oelewapperke | unless I can "originate" on an existing channel |
| 12:02.03 | leifmadsen | oelewapperke: sounds like you want to use the CLI originate command |
| 12:02.40 | Exten | <PROTECTED> |
| 12:04.51 | kaldemar | oelewapperke: how is there an existing channel if you have no dialplan? |
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| 12:07.28 | kaldemar | Exten: nothing. what makes you think something is? |
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| 12:09.55 | Exten | [May 3 14:59:47] ERROR[11643]: app_exec.c:191 execif_exec: Invalid Syntax. |
| 12:10.07 | wdoekes2 | it's a bit verbose perhaps |
| 12:10.10 | wdoekes2 | Set(PGENDER=${IF($["${PGENDER}"="1"]?M:F)}) |
| 12:10.45 | Amorsen | Nicely done wdoekes2, I was going for the same thing but got lost in crunchy brackets |
| 12:10.55 | wdoekes2 | or are you using old asterisk? (1.4) |
| 12:11.09 | wdoekes2 | in which case you need comma's instead of parentheses |
| 12:11.15 | kaldemar | Exten: what version are you using? the exact line you pasted works in 1.8.3.3. |
| 12:11.16 | Exten | 1.4 |
| 12:11.30 | wdoekes2 | Set,PGENDER=M:... |
| 12:11.56 | kaldemar | Set() works fine in 1.4, even in 1.2. |
| 12:12.01 | wdoekes2 | possibly even a , where the first ? is |
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| 12:12.35 | Exten | _X.,n,ExecIf($[ "${PGENDER}" = "1" ],Set(PGENDER=M),Set(PGENDER=F)) <-- ? |
| 12:12.42 | wdoekes2 | no |
| 12:12.49 | wdoekes2 | more comma's |
| 12:13.44 | wdoekes2 | exten => closed,n,ExecIf($["${vm_exten}"!=""&"${af_closed}"=""],VoiceMail,${vm_exten}@${customercode}) |
| 12:13.57 | oelewapperke | kaldemar: because I'm using async AGI |
| 12:14.04 | Exten | ill try the set thingie |
| 12:14.17 | kaldemar | Exten: try ExecIf($["${PGENDER}" = "1"],Set(PGENDER=M),Set(PGENDER=F)) |
| 12:14.22 | jacc0 | @Exten: what version are you using? |
| 12:14.27 | Exten | 1.4 |
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| 12:15.02 | kaldemar | eh, no. ExecIf is pretty different in 1.4, ExecIF (<expr>|<app>|<data>). |
| 12:15.46 | wdoekes2 | kaldemar: my ExecIf example works.. but I don't think you get the Else syntax |
| 12:15.49 | kaldemar | no possibility to execute something on false. |
| 12:15.58 | Exten | if thingie works ! |
| 12:17.27 | kaldemar | wdoekes2: else syntax as in the 1.4 app? |
| 12:18.20 | wdoekes2 | "the Else syntax" == "no possibility to execute something on false" |
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| 12:22.36 | leifmadsen | on 1.4 you have to be clever |
| 12:23.01 | leifmadsen | exten => start,1,Exec(${IF($[${foo} = ${bar}]?Voicemail(100@default):NoOp())}) |
| 12:23.59 | leifmadsen | because there is no "else" on ExecIf(). It's either, "Execute this if true, or don't do anything" |
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| 12:51.21 | oelewapperke | is there some way via AMI to get an existing channel into the dialplan ? |
| 12:51.54 | oelewapperke | Originate + Bridge is very complex to control, and it seems wasteful, creating 2 new channels just for this ? |
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| 14:21.30 | nokiz | Hi everybody, I have a problem thats been killing me for 2 days now. Got soft phones running just fine but when connecting a Polycom SoundStation IP 5000 all i get it "ACL error (permit/deny)" when it tries to register and a really dont understand why, as far as i can tell sip.conf is correct. Ive been searched my a** off but no result. Any one got any suggestions or can help me? |
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| 14:24.22 | leifmadsen | nokiz: sounds like you've got something set via permit/deny options in sip.conf somewhere |
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| 14:25.49 | nokiz | leifmadsen: only thing in the sip.conf file exept for the sip accounts are [general] useragent=Cokomm CPBX bindaddr=0.0.0.0 insecure=very srvlookup=no context=default-custom language=de tos=0x18 maxexpirey=3600 defaultexpirey=3600 dtmfmode=auto disallow=all allow=alaw allow=ulaw allow=g726 allow=gsm allow=g729 alwaysauthreject=yes |
| 14:26.16 | nokiz | cant find any permit/deny options. Should i add them |
| 14:26.18 | nokiz | ?? |
| 14:27.56 | kaldemar | nokiz: do you have deny options under the peer that the client matches to? |
| 14:28.29 | *** join/#asterisk sereal (~jjrh@2001:4830:16ca:1:21a:6bff:fe6a:3cd2) |
| 14:29.51 | nokiz | kaldemar: As far as i can tell there is none, However i did not install the system so im not surtain over all the settings. |
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| 14:37.00 | kaldemar | nokiz: look for #include's in the sip.conf. also check in extconfig.conf that realtime is not used. |
| 14:38.11 | sereal | If the directory application uses the voicemail.conf settings to create the list how does one deal with searching users by last name - it doesn't look like you can create a sip account with a space in the name. |
| 14:38.32 | sereal | Is there a alternative way to define a set of names and extensions to create a directory? |
| 14:39.09 | nokiz | kaldemar: no #includes in sip.conf and extconfig.conf do not have and is not in use |
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| 14:55.55 | cusco_ | hey folks |
| 14:56.17 | cusco_ | moh class has files but I do not understand the order... |
| 14:56.20 | cusco_ | http://paste.debian.net/115883/ |
| 14:56.27 | cusco_ | could some one help me out? |
| 14:56.37 | cusco_ | it goes moh3 then moh1 then moh2 |
| 15:00.24 | cusco_ | ok I solverd it |
| 15:01.15 | cusco_ | I put moh1, then moh reload, put moh2, moh reload and put moh3 and moh reload |
| 15:02.49 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 15:02.55 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 15:09.51 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 15:11.23 | *** join/#asterisk Tier3TechLS (~lsaldivar@184.18.157.26) |
| 15:11.30 | Tier3TechLS | Hello |
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| 15:12.15 | Aut0ExeC | Tier3TechLS: sup man |
| 15:12.21 | Tier3TechLS | NM hows it goin? |
| 15:12.47 | Aut0ExeC | <PROTECTED> |
| 15:12.51 | Aut0ExeC | havent seen u around |
| 15:13.01 | Tier3TechLS | Nope just had my cherry popped |
| 15:13.02 | Tier3TechLS | ha |
| 15:13.09 | Aut0ExeC | ahh ok |
| 15:13.22 | sxpert | that has interesting meanings in some areas |
| 15:13.30 | Tier3TechLS | LOL |
| 15:13.32 | Tier3TechLS | yes it does |
| 15:13.33 | Aut0ExeC | by "cherry popped" u mean opened ur can of dr pepper cherry right? |
| 15:13.42 | Tier3TechLS | yes sir! lol |
| 15:13.46 | Aut0ExeC | ok ok |
| 15:13.47 | Aut0ExeC | :) |
| 15:14.18 | Tier3TechLS | Im just getting into the VOiP area and learning as much as possible. |
| 15:14.33 | Tier3TechLS | Always used a hosted solution. I want to build my own. |
| 15:14.47 | Aut0ExeC | ahh ok |
| 15:14.52 | Aut0ExeC | i kind of a newbie too |
| 15:14.56 | Tier3TechLS | Paying 60bucks for 1 hosted line isnt something i want to do anymore. |
| 15:15.04 | Aut0ExeC | yah sux |
| 15:15.17 | Aut0ExeC | i just setup asterisk on my linksys router |
| 15:15.20 | Tier3TechLS | I just installed AsteriskNow with FreePBX. |
| 15:15.22 | Aut0ExeC | it rox basically |
| 15:15.28 | Aut0ExeC | freepbx ? |
| 15:15.32 | Aut0ExeC | ahh ok |
| 15:15.34 | sxpert | did interesting multi-people chatrooms for the cherry-popping community |
| 15:15.37 | Tier3TechLS | I have 3 Aastra phones also. |
| 15:15.42 | Tier3TechLS | LOL |
| 15:15.51 | Tier3TechLS | Ill stick to this one for now. |
| 15:15.56 | sereal | you can run asterisk on a linksys router? |
| 15:15.57 | Aut0ExeC | sxpert: we might be off the cherry popping now |
| 15:15.58 | sxpert | interesting in the software, not for whatever they talk about in there, obviously |
| 15:16.07 | Aut0ExeC | sereal: lol i just said yes |
| 15:16.16 | sereal | what linksys? |
| 15:16.29 | sxpert | linksys actually does routers ? |
| 15:16.29 | Aut0ExeC | any router thats compatilbe with openwrt |
| 15:16.37 | Aut0ExeC | well a fairly decent one |
| 15:16.39 | Aut0ExeC | with good specs |
| 15:16.43 | Aut0ExeC | i use wrt54gl |
| 15:16.45 | sereal | what model are you doing this with? |
| 15:16.48 | sxpert | I thought they were just doing crappy switches ;) |
| 15:16.49 | sereal | ah |
| 15:17.02 | Aut0ExeC | with sd card mod |
| 15:19.02 | Tier3TechLS | Are there any guides or walkthroughs for Asterisk? |
| 15:19.26 | Tier3TechLS | I have it all installed. I am having trouble with the phone connecting to the PBX. |
| 15:19.32 | gruvfunk | There's the "book" |
| 15:19.36 | Tier3TechLS | LOL |
| 15:19.48 | sereal | or the asterisk webpage |
| 15:19.55 | leifmadsen | or google :) |
| 15:19.57 | gruvfunk | the wiki |
| 15:20.08 | sxpert | the wiki is pretty decent |
| 15:20.10 | sereal | like google asterisk tutorial |
| 15:20.14 | wdoekes2 | ~newbook |
| 15:20.15 | infobot | Please see ~thebook for more information about Asterisk: The Definitive Guide |
| 15:20.21 | wdoekes2 | ~thebook |
| 15:20.21 | 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 |
| 15:20.25 | m4xx | has anyone gotten sphinx or pocketsphinx to work with asterisk 1.8? |
| 15:20.29 | *** join/#asterisk nokiz (d5a5a512@gateway/web/freenode/ip.213.165.165.18) |
| 15:20.39 | gruvfunk | i can never remember the shortcut :) |
| 15:20.48 | sxpert | has fring working |
| 15:21.06 | m4xx | i'm more looking for the speech recognition part |
| 15:21.20 | sereal | If the directory application uses the voicemail.conf settings to create the list how does one deal with searching users by last name - it doesn't look like you can create a sip account with a space in the name. |
| 15:22.02 | Tier3TechLS | Ill have to check all that out. Im using Asterisk now. Its all GUI. |
| 15:22.09 | m4xx | as for tts cepstral david sound _a lot_ better to me |
| 15:22.34 | nokiz | Does anyone know more about this problem other than what kaldemar and leifmadsen said?: NOTICE[12258]: chan_sip.c:11529 handle_request_register: Registration from '<sip:192.168.1.xx@192.168.1.xx>' failed for '192.168.1.xx' - ACL error (permit/deny) |
| 15:22.43 | leifmadsen | m4xx: I got it working, but it doesn't parse words worth a shit |
| 15:23.00 | leifmadsen | it is basically useless |
| 15:23.15 | m4xx | I'm trying to interact with an IVR |
| 15:23.24 | m4xx | do you think it would be as usless to me? |
| 15:23.24 | leifmadsen | yes I understand what you're trying to do |
| 15:23.44 | leifmadsen | if it doesn't recognize your words, then yes, I presume it would be useless |
| 15:23.51 | m4xx | lol |
| 15:23.53 | leifmadsen | you will have to learn how to tune sphinx |
| 15:24.25 | Tier3TechLS | Anyone using AsteriskNOW? |
| 15:24.38 | leifmadsen | probably lots of people |
| 15:24.42 | m4xx | Tier3TechLS, but the book and save yourself the headaches ;D |
| 15:24.47 | sxpert | http://asset.rue89.com/files/ColinVerot/head_shot.jpg |
| 15:24.54 | m4xx | learn how to do it by hand then try the gui's |
| 15:25.15 | m4xx | *buy the book |
| 15:25.21 | sereal | ^ |
| 15:25.35 | sereal | I believe the book is free as a ebook |
| 15:25.38 | Aut0ExeC | doesnt freepbx have completely diff configs? |
| 15:25.44 | leifmadsen | yes |
| 15:25.49 | leifmadsen | it generates the configuration files |
| 15:25.52 | Aut0ExeC | i dont think the book covers that |
| 15:25.57 | leifmadsen | it certainly doesn't :) |
| 15:26.02 | Aut0ExeC | so that wont help him |
| 15:26.10 | Aut0ExeC | perhaps a freepbx book |
| 15:26.11 | Aut0ExeC | ? |
| 15:26.13 | leifmadsen | it will help him to learn how to read dialplan |
| 15:26.30 | sereal | show dialplan |
| 15:26.37 | leifmadsen | dialplan show |
| 15:26.40 | m4xx | leif, how long have you tried it? did you use the 8000hz model? |
| 15:26.40 | Aut0ExeC | lol |
| 15:26.55 | Aut0ExeC | no diaplan specified |
| 15:26.56 | leifmadsen | m4xx: yes I played around -- I followed the same post you're probably reading |
| 15:29.50 | Aut0ExeC | sxpert: u think obama really plays ps3? |
| 15:30.30 | Aut0ExeC | headshot pic is funny... he looks so focused |
| 15:30.32 | Tier3TechLS | Is the GlobalSIP for the local PBX? |
| 15:31.13 | Aut0ExeC | Tier3TechLS: you should start from the basics bro |
| 15:31.19 | Aut0ExeC | Tier3TechLS: get the book like they suggested |
| 15:31.48 | *** join/#asterisk remnant13 (~Gray@unaffiliated/remnant13) |
| 15:31.57 | sxpert | Aut0ExeC: that's obama playing the most expensive counterstrike game ever ;) |
| 15:32.06 | *** part/#asterisk Tier3TechLS (~lsaldivar@184.18.157.26) |
| 15:32.07 | Aut0ExeC | sxpert: lol |
| 15:32.12 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 15:32.32 | sxpert | Aut0ExeC: aka hunting OBL down |
| 15:32.49 | Aut0ExeC | lol |
| 15:33.13 | Aut0ExeC | guess he won |
| 15:33.22 | sxpert | yeah |
| 15:33.33 | Aut0ExeC | navy seals rock man |
| 15:33.40 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:34.11 | sxpert | or so he says. can't verify it though, body is supposed to be feeding fishes at the bottom of the indian ocean |
| 15:34.38 | Aut0ExeC | exactly... they have pics tho.. to release soon |
| 15:34.59 | Aut0ExeC | u can verify |
| 15:35.00 | Aut0ExeC | :) |
| 15:35.24 | sxpert | haven't seen any yet |
| 15:35.36 | sxpert | waiting for wikileaks ;) |
| 15:36.13 | Aut0ExeC | ya, only a matter of time |
| 15:39.50 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-130-106.twcny.res.rr.com) |
| 15:39.58 | m4xx | i'm trying to interact with someone elses ivr. I've sort of got it working but some times there's an anouncement that i can't really account for. i wanted to use the voice recognition for this but can't really afford the lumenvox app at the moment. I just thought of a hack that would be to record the prompts and get a slice of the raw data hash it. If it matches a known hash then continue. |
| 15:39.58 | m4xx | Does this come off as unrealistic to anyone? |
| 15:40.47 | m4xx | or can anyone suggest a better solution? |
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| 15:58.25 | Besticles | I'm having an issue with Cepstral. I'm making a call out using Local & DAHDI, I stream a file, then say the persons name via swift. It works if I make one call out. It works if I make 4 calls out the same time. If I make 8 calls out, 6 work, the other 2 starts returning: |
| 15:58.25 | Besticles | [May 3 03:01:04] WARNING[26575]: channel.c:1044 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/apexoutbound@apexdial-9126;1 |
| 15:59.10 | Besticles | I don't know where to begin to fix this issue. I know it's cepstral, because I commented out Swift, and made 50 simulataneous calls and no errors. |
| 15:59.55 | Besticles | Does anyone have any pointers how to troubleshoot the Exceptionally long voice queue errors? |
| 16:00.27 | sxpert | never seen that |
| 16:00.52 | ectospasm | may need to file a bug with Cepstral... |
| 16:01.04 | *** join/#asterisk draeath (~draeath@unaffiliated/draeath) |
| 16:03.20 | tzanger | Cepstral sounds like a cough medicine to me |
| 16:03.31 | tzanger | which is probably not a good connection to speech software |
| 16:03.44 | Besticles | Already submitted a ticket :P Thought I would ask over here too. Thanks for your 2 cents. |
| 16:04.11 | coppice | Cepstral is too technical a name for most people to relate to |
| 16:04.55 | coppice | heck, most people aren't sure how to pronounce it :-) |
| 16:05.22 | m4xx | it's better than any of the free solutions in my opinion, and far better price than any alternative =[ |
| 16:06.23 | *** join/#asterisk fish-bulb (~qcstewart@nat/digium/x-smrerrpamucbgoei) |
| 16:07.32 | gruvfunk | hey all, really starting to get frustrated with providers who say they support Asterisk, provide you with a sample configuration and then refuse to provide "device support" to troubleshoot why it's not working |
| 16:08.06 | gruvfunk | they suggest type=friend - is this still viable in *1.8 ? |
| 16:08.26 | ectospasm | gruvfunk: I believe so... |
| 16:08.42 | gruvfunk | friend = peer + user, yeah? |
| 16:09.01 | ectospasm | yes. A user can send Asterisk calls, Asterisk sends peers calls |
| 16:09.29 | ectospasm | friends are a shortcut to do both |
| 16:09.44 | oelewapperke | gruvfunk: there's a good reason for that : try to be on the other end of that support for once |
| 16:11.23 | gruvfunk | oelewapperke: agree it's painful, been there done it - however, I don't think they should provide customers with sample "working" configs if they're not going to stand up behind them |
| 16:12.20 | leifmadsen | it costs more money to support an end-user than you can possibly make from them in a year |
| 16:12.55 | gruvfunk | in this case it's a Dr's office account with several DID's and tens of thousands of minutes per month |
| 16:13.05 | gruvfunk | 1 DID on *1.8 not working |
| 16:13.20 | leifmadsen | then maybe it's on their end |
| 16:13.26 | gruvfunk | right, I've suggested |
| 16:13.41 | leifmadsen | perhaps you can provide some information that allows us to help? |
| 16:13.58 | gruvfunk | they did a sip trace and tell me that we're not responding to their ACK after authentication and media stream, we flood them with SIP200 OK |
| 16:14.14 | gruvfunk | I'd love to leifmadsen, where do I begin |
| 16:14.15 | oelewapperke | gruvfunk: do you have a sip trace yourself ? |
| 16:14.43 | oelewapperke | just start tcpdump -w log.pcap -s 9999 -i <incoming interface> and call whatever did is not working, ctrl-c and upload the file somewhere |
| 16:14.51 | leifmadsen | ~collectdebug |
| 16:14.51 | infobot | collectdebug is probably a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 16:14.57 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 16:15.11 | gruvfunk | truly awesome |
| 16:15.41 | gruvfunk | be back with data |
| 16:22.10 | KavanS | uggh - I'm stuck on asterisk 1.4.32 and I can't seem to get the buddylist feature working on these polycom's |
| 16:22.35 | KavanS | anyone have any success with asterisk 1.4 and this feature? - I've even patched chan_sip.c as directed by a few guides online |
| 16:23.01 | Chainsaw | Wow, 1.4; I'd try on 1.6.2 at the very least. |
| 16:23.14 | KavanS | uggh...not the most exciting news I wanted to hear |
| 16:23.18 | KavanS | ok... |
| 16:23.32 | Chainsaw | I have had it working on 1.6.2; I never did run 1.4 so I can't help you with that one. |
| 16:23.45 | Chainsaw | went from 1.2 to 1.6.0 (and yes, that was excruciatingly painful) |
| 16:24.04 | KavanS | yeah I'm concerned about the differences from 1.4 to 1.6.... |
| 16:24.12 | KavanS | am I going to have to hack up my dialplan to make things work? |
| 16:24.30 | Chainsaw | Quite likely. You will have this pain now or down the road. Do you have a spare box you can test on? |
| 16:24.37 | *** join/#asterisk felimwhiteley (~quassel@109.255.104.145) |
| 16:24.41 | KavanS | yep, I've got some spare hardware |
| 16:24.59 | Chainsaw | I would say build up a box, take your 1.4 config and hack it up until you think it works. |
| 16:25.16 | sxpert | man, there are still 1.4 installations ? |
| 16:25.16 | KavanS | sxpert, lol hell yes there are... |
| 16:25.16 | Chainsaw | Switch over live traffic for a bit and do the fine tuning. |
| 16:26.41 | KavanS | ok...fun, project ahead |
| 16:26.46 | KavanS | any words about custom agi scripts? |
| 16:32.36 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v002-031.mobile.uci.edu) |
| 16:33.56 | *** join/#asterisk manji (~manjiki@ppp-2-84-9-127.home.otenet.gr) |
| 16:34.16 | Chainsaw | KavanS: I don't use those, just PHP scripts driving AMI. |
| 16:34.20 | Chainsaw | KavanS: So I can't comment. |
| 16:34.26 | KavanS | ok right on |
| 16:37.31 | gruvfunk | leifmadsen: took the logger approadh, almost ready but questioning... why am I seeing so many ^M characters ending lines? |
| 16:37.32 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:37.42 | gruvfunk | s/approadh/approach |
| 16:38.57 | *** join/#asterisk nix8n82 (~nate@24.143.28.16) |
| 16:41.15 | leifmadsen | gruvfunk: because you're logging with colour enabled, so you're seeing ANSI chars |
| 16:42.01 | gruvfunk | scp'ed the file over to notepad in windows and no longer see those ^M |
| 16:42.43 | gruvfunk | pastebin the sip trace? |
| 16:46.47 | gruvfunk | leifmadsen and oelewapperke http://paste2.org/p/1397148 |
| 16:47.51 | oelewapperke | is it "allowed" to Originate to "Local/exten@context", and then execute AGI on the other end of that link ? |
| 16:48.46 | cyford | hi, is there anyway i can capture the agent name to a variable from ${BLINDTRANSFER} |
| 16:49.03 | cyford | in dialplan |
| 16:56.52 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:56.52 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 16:58.26 | cyford | (rewording) Can Set(foo=${DB(family/key)}) return sip names based on extenstion |
| 16:59.48 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 16:59.48 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:59.55 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 16:59.57 | dan__t | So a channel being a single connection in to/out of Asterisk, a "line", I get that part. What's it called when multiple channels are combined, such in the case of essentially two people having a conversation? Is that a bridge? |
| 17:00.38 | pabelanger | dan__t: usually, yes |
| 17:01.33 | dan__t | Ok, good. I was playing around with Monitor(), and the results were only the calling party's channel audio being recorded. I combined the two streams but I understand then to be "input" and "output", not a combined stream session of both channels being bridged, as I kind of expected. |
| 17:03.57 | dan__t | I'm guessing Monitor() is not what I'm looking for. If I wanted to record the call in its entirety, including all channels on that bridge, what would I be looking for? |
| 17:05.48 | nix8n82 | cyford, yes if you set up the family/key to hold such values |
| 17:06.23 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 17:06.56 | leifmadsen | dan__t: yes two channels get "bridged" in order to have a "call" |
| 17:07.06 | dan__t | right |
| 17:07.07 | *** join/#asterisk mawhiii (~trav@tcmsdev.com) |
| 17:07.22 | dan__t | Pretty sure I recall (and understand) that from reading your book, actually. |
| 17:08.01 | gruvfunk | leifmadsen and oelewapperke or anyone else - please ping directly if you find anything in the sip trace http://paste2.org/p/1397148 |
| 17:08.08 | leifmadsen | dan__t: ok knowing that, Monitor() (or MixMonitor()) will record all conversations/bridges that the channel that executed the dialplan application |
| 17:08.34 | gruvfunk | yesterday I was getting 32 seconds of call time, today I'm down to 6seconds (which is what we experienced in weeks past with this provider) |
| 17:08.35 | dan__t | The channel that executed the dialplan.... ok, so the calling party executed the call. |
| 17:08.43 | dan__t | Monitor() will only record on that channel. |
| 17:10.28 | leifmadsen | use MixMonitor() and it records both channels |
| 17:11.02 | leifmadsen | dan__t: just remember that the recording follows the channel that executed the dialplan, so when you do things like transfers and such, remember that it will follow that channel, and stop recording if that channel hangs up or is destroyed |
| 17:11.22 | leifmadsen | so in a transfer type scenario your recording may stop if the calling channel performs a transfer and hangs up |
| 17:11.53 | *** join/#asterisk \DSAFEW\ (~DSAFEW_@ip72-208-176-219.ph.ph.cox.net) |
| 17:11.58 | dan__t | MixMonitor(${UNIQUEID},b) |
| 17:12.10 | dan__t | Ahhh, got it. |
| 17:12.23 | dan__t | but then I should be able to append to that very same file, with the a option |
| 17:12.30 | dan__t | I mean that's what a smart person would do |
| 17:12.40 | *** join/#asterisk Sorcier_FXK (~Sorcier_F@unaffiliated/sorcierfxk) |
| 17:19.30 | *** join/#asterisk cerberus_za (~coert@196-215-29-217.dynamic.isadsl.co.za) |
| 17:23.08 | gruvfunk | interesting, my outbound termination is just fine, I can call out and the call stays up |
| 17:23.28 | gruvfunk | but inbound origination is not holding up, this has to be a sip.conf thing, right? |
| 17:24.22 | Tozz_ | or dialplan |
| 17:25.16 | *** join/#asterisk Jcook_5xData (~Jcook_5xD@173.162.32.1) |
| 17:25.47 | *** join/#asterisk c2tarun (~quassel@1.23.170.246) |
| 17:25.51 | Jcook_5xData | anyone here use astassistant? |
| 17:26.40 | leifmadsen | never heard of it |
| 17:27.34 | gruvfunk | i'm posting my sip.conf --> http://paste2.org/p/1397213 |
| 17:27.41 | Jcook_5xData | http://www.astassistant.com/ |
| 17:27.45 | c2tarun | I want to desing an application in Qt that can as many aspects of Asterisk as possible. For that I need to know bit more information of how asterisk work and how to implement applications in asterisk framework. Can anyone please guide me to any reference material that can help me. I never did this before |
| 17:27.54 | gruvfunk | please let me know if you can help - if indeed something is wrong on my side - going for a bite to eat |
| 17:28.02 | c2tarun | I want to desing an application in Qt that can manage as many aspects of Asterisk as possible. For that I need to know bit more information of how asterisk work and how to implement applications in asterisk framework. Can anyone please guide me to any reference material that can help me. I never did this before |
| 17:28.42 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 17:29.04 | Jcook_5xData | c2tarun, You can look at http://www.astassistant.com/ I think it opensource I think it does the same as what you are looking to |
| 17:34.41 | c2tarun | Jcook_5xData: thats cool :) but astassistant is an application. true I want to design an application similar to it, but that requires me to understand asterisk first. |
| 17:35.04 | dan__t | Thanks for explaining, leifmadsen. |
| 17:35.39 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:36.47 | c2tarun | Jcook_5xData: can you guide me to a reference material that can help me in understanding asterisk first? |
| 17:36.59 | ectospasm | ~thebook |
| 17:36.59 | 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 |
| 17:38.40 | ectospasm | c2tarun: ^ |
| 17:40.37 | c2tarun | ectospasm: sorry but that is completely an asterisk guide :( nothing mentioned of how to access the functionalities of asterisk with code. hope you understanding what I am trying to ask? |
| 17:42.00 | _Corey_ | c2tarun: Google Asterisk AGI |
| 17:42.08 | _Corey_ | c2tarun: or Asterisk Manager |
| 17:43.50 | c2tarun | _Corey_: I think this will work :) thanks a lot mate, you are a lifesaver |
| 17:44.01 | _Corey_ | sure |
| 17:44.47 | leifmadsen | c2tarun: read the chapters on AGI and AMI |
| 17:44.58 | leifmadsen | c2tarun: it most certainly does talk about what you're looking for :) |
| 17:45.18 | c2tarun | leifmadsen: chapters as in this http://ofps.oreilly.com/titles/9780596517342/ book? |
| 17:45.29 | leifmadsen | yes |
| 17:45.39 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 17:45.40 | leifmadsen | if you're looking for information about the architecture of asterisk, read this: http://www.aosabook.org/ |
| 17:45.42 | c2tarun | leifmadsen: got it :) thanks |
| 17:45.43 | leifmadsen | first chapter |
| 17:46.31 | Jcook_5xData | what would this mean "but no invalid handler"? |
| 17:46.35 | c2tarun | leifmadsen: well this may be too much of asking, but do you think that knowledge about AGI or AMI will help me completely or I need to read something else too? |
| 17:46.52 | leifmadsen | c2tarun: not knowing what you're doing... yes |
| 17:47.06 | leifmadsen | Jcook_5xData: no 'i' extension to handle invalid extensions |
| 17:47.22 | leifmadsen | Jcook_5xData: check the dialplan chapters in the book mentioned above |
| 17:48.59 | Jcook_5xData | leifmadsen, thanks |
| 17:51.51 | cusco_ | where can I read about setting qos for sip ? :P |
| 17:56.28 | wonderworld | hi guys, i am trying to setup a meetme conference on my asterisk. when i user tries to join the conference, i get "app_meetme.c:1097 build_conf: Unable to open pseudo device" |
| 17:57.11 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 17:57.21 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 17:59.51 | *** join/#asterisk Polysics (~Luca@host51-72-dynamic.41-79-r.retail.telecomitalia.it) |
| 17:59.55 | Polysics | hello |
| 18:00.06 | Polysics | would you use 1.8.3.3 for a fresh installation? |
| 18:00.15 | Polysics | 1.8, basically |
| 18:01.24 | gruvfunk | Polysics: I do |
| 18:01.37 | Polysics | i need to use adhearsion with it but i suppose that is not an issue |
| 18:01.47 | Polysics | since 1.0.1 does mention 1.8 |
| 18:01.49 | Polysics | ok, thanks |
| 18:01.53 | Polysics | later i will install it :-9 |
| 18:03.52 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-130-106.twcny.res.rr.com) |
| 18:06.51 | gruvfunk | 234 users: anyone using Gafachi provider (Rochester, NY) |
| 18:08.59 | malcolmd | nope |
| 18:09.19 | *** join/#asterisk Freeaqingme_ (~dolf@dsl-083-247-011-232.solcon.nl) |
| 18:09.22 | malcolmd | wonderworld: you don't have dahdi loaded |
| 18:12.25 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-130-106.twcny.res.rr.com) |
| 18:13.28 | gruvfunk | chan_sip.c:3511 retrans_pkt: Retransmission timeout reached on transmission 429f089d465e8b040693f2eb5ba36c56@67.216.35.162 for seqno 103 (Critical Response) ; Packet timed out after 6400ms with no response ; Hanging up call 429f089d465e8b040693f2eb5ba36c56@67.216.35.162 - no reply to our critical packet |
| 18:13.29 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 18:14.09 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 18:17.28 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 18:19.30 | wonderworld | malcolmd: yes, thanks. fixed it |
| 18:19.40 | malcolmd | cool |
| 18:20.32 | *** join/#asterisk trijezdci (~trijezdci@84-231-18-132.elisa-mobile.fi) |
| 18:22.29 | *** join/#asterisk tiz- (wce@97.103.4.72) |
| 18:23.36 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 18:24.53 | *** join/#asterisk JonnyD_work (~Jon@173.226.80.154) |
| 18:25.30 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 18:27.17 | *** join/#asterisk devil_evoxxx (~d3v1l@host137-92-dynamic.0-79-r.retail.telecomitalia.it) |
| 18:27.27 | devil_evoxxx | hi guys |
| 18:30.40 | devil_evoxxx | i have an asterisk 1.4.37 and i want to install MeetMe |
| 18:30.56 | devil_evoxxx | is really necessary to install dahdi? |
| 18:31.05 | devil_evoxxx | (i can't select on menuselect of asterisk) |
| 18:32.12 | *** join/#asterisk binbash_ (~peter@a80-127-249-195.adsl.xs4all.nl) |
| 18:32.22 | rogersja | devil_evoxxx: have you compiled dahdi? |
| 18:32.29 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-130-106.twcny.res.rr.com) |
| 18:34.47 | kaldemar | devil_evoxxx: meetme requires dahdi. you can't select it in the menu of asterisk, it is a separate package. or two pckages actually. |
| 18:35.27 | wonderworld | devil_evoxxx: i just did it, it's easy. get dahdi, unpack, make, make install, modprobe dahdi, restart asterisk |
| 18:36.31 | wonderworld | you might also need to get the headers for your current kernel first |
| 18:39.12 | *** join/#asterisk ketema (~ketema@2001:470:5:138:219:e3ff:fe09:5b25) |
| 18:39.54 | *** part/#asterisk ketema (~ketema@2001:470:5:138:219:e3ff:fe09:5b25) |
| 18:42.16 | Aut0ExeC | do you really need dahdi to use meetme? |
| 18:42.18 | Aut0ExeC | weird |
| 18:43.11 | pabelanger | Aut0ExeC: yes |
| 18:43.23 | pabelanger | otherwise use conf_bridge |
| 18:43.29 | pabelanger | err... ConfBridge() |
| 18:44.09 | Aut0ExeC | k |
| 18:45.42 | anonymouz666 | AFAIK, the DAHDI itself is the responsible to mix the audio, not the MeetMe application. |
| 18:46.12 | *** join/#asterisk m_tadeu (~quassel@89.180.67.188) |
| 18:46.34 | rogersja | anonymouz666: doesn't dahdi simply provide a timing source for meetme? |
| 18:47.16 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 18:47.24 | Aut0ExeC | anonymouz666: ahhh ok... good info |
| 18:50.18 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:50.18 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:50.18 | *** join/#asterisk rogersja (~RogersJa@S0106000f6695039f.gv.shawcable.net) |
| 18:52.23 | leifmadsen | the whole point of MeetMe() is to mix audio :) |
| 18:52.34 | leifmadsen | DAHDI provides the timing to know how much to mix (as far as I understand it) |
| 18:52.44 | leifmadsen | russellb: would be best to explain how it works |
| 18:53.31 | anonymouz666 | leifmadsen: the meetme does not use the audio in slin provided by dahdi? I remember reading a mail with kpfleming explaining the mix was done in dahdi |
| 18:53.43 | leifmadsen | no idea |
| 18:53.50 | *** join/#asterisk vfabi (~fabi@194.247.164.231) |
| 18:53.59 | rogersja | i thought that dahdi just provided timing as well |
| 18:54.17 | anonymouz666 | http://lists.digium.com/pipermail/asterisk-users/2008-July/215444.html |
| 18:54.41 | *** join/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
| 18:55.04 | *** part/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
| 18:55.34 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 18:58.36 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 19:03.36 | p3nguin | Is a PAP2T suitable for a security alarm system? Our alarm panel requires "regular dial tone service," according to the alarm company. It simply calls a toll free number in the event of an unauthorized entry, so I don't see any real difference between using land line service and using a combination of Asterisk, an ITSP, and a PAP2T to give it a dial tone. Anyone else here run an alarm through Asterisk? |
| 19:03.46 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 19:04.21 | anonymouz666 | yes |
| 19:07.45 | anonymouz666 | PAP2T - asterisk - E1 - PSTN |
| 19:08.13 | anonymouz666 | my only worry is not allow asterisk to touch my DTMFs duration. |
| 19:08.42 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-130-106.twcny.res.rr.com) |
| 19:09.14 | devil_evoxxx | thankypi still having this problem now with meetme |
| 19:09.17 | devil_evoxxx | [May 3 21:03:13] WARNING[19470]: app_meetme.c:829 build_conf: Unable to open pseudo device -- <SIP/16-00000002> Playing 'conf-invalid' (language 'it') |
| 19:09.53 | anonymouz666 | pseudo device I think it's DAHDI stuff I was talking about |
| 19:10.38 | devil_evoxxx | dahdi is now installed |
| 19:10.38 | devil_evoxxx | but |
| 19:11.01 | devil_evoxxx | i don't need a conference room on dahdi channels, i need a conference room only trought sip phones |
| 19:11.14 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 19:11.56 | anonymouz666 | you didn't understand the point |
| 19:13.30 | anonymouz666 | meetme uses the dahdi pseudo channels |
| 19:13.33 | devil_evoxxx | now yes, ( google miracles)..i found a post where they say that change "/dev/zap/pseudo |
| 19:13.33 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 19:13.47 | devil_evoxxx | change owner..but |
| 19:13.56 | devil_evoxxx | i have not pseudo device..why? |
| 19:14.09 | anonymouz666 | what dahdi version? |
| 19:15.20 | devil_evoxxx | the last 2.4.1.2 |
| 19:15.29 | devil_evoxxx | and 1.4.37 asterisk version |
| 19:15.50 | anonymouz666 | make sure you modprobe dahdi |
| 19:16.09 | anonymouz666 | I guess this version does not require dahdi_dummy anymore |
| 19:16.12 | ruben23 | hi guys what version of asterisk best support H323 for video call..any idea |
| 19:16.41 | leifmadsen | ruben23: 1.8 and greater using chan_ooh323 |
| 19:16.52 | p3nguin | In the later versions of Dahdi, the dummy is built in. |
| 19:17.19 | anonymouz666 | p3nguin: this is nice |
| 19:17.49 | JonnyD_work | does anyone know of a cloud hosting providor that lets people use their own iso? |
| 19:18.54 | devil_evoxxx | anonymouz, thankyou..now work fine! |
| 19:19.16 | anonymouz666 | devil_evoxxx: happy MeetMe() |
| 19:20.12 | Aut0ExeC | where do I go to fix the date beign logged in cdr? |
| 19:20.13 | p3nguin | If I can use a single PAP2T for the alarm panel and for a fax machine, that would be even better. |
| 19:20.53 | Aut0ExeC | date on my system is right but cdr is logging a bad time |
| 19:21.04 | ruben23 | leifmadsen: how about asterisk 1.6 ver...? |
| 19:21.14 | leifmadsen | ruben23: my answer still stands |
| 19:21.29 | leifmadsen | ruben23: 1.8 has many fixes that provides a working H323 channel driver |
| 19:21.36 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 19:21.39 | leifmadsen | you can try something older, but you won't get the same results |
| 19:22.04 | leifmadsen | only chan_ooh323 has been actively worked on in quite some time, and the good changes are in 1.8 |
| 19:22.09 | leifmadsen | so you must use that |
| 19:22.19 | leifmadsen | anything else is a crap shoot |
| 19:23.12 | anonymouz666 | ruben23: just out of curiosity, what are you using that requires h323? |
| 19:23.33 | Aut0ExeC | leifmadsen: where do I go to sync my cdr time with my system time? |
| 19:23.58 | leifmadsen | NTP? |
| 19:24.02 | cusco_ | o.O cdr should use system time |
| 19:24.12 | ruben23 | <PROTECTED> |
| 19:24.17 | Aut0ExeC | leifmadsen: my system time is correct |
| 19:24.18 | cusco_ | asterisk's ${EPOCH} is system's time |
| 19:24.32 | Aut0ExeC | cdr time is off |
| 19:24.34 | Aut0ExeC | by 3 hours |
| 19:25.19 | ferdna | guys, i am getting error 415 in my grandstream BT202... i can receive calls perfectly... but cannot make outgoing calls |
| 19:25.21 | ferdna | any ideas? |
| 19:25.35 | ferdna | my voip provider is broadvoice |
| 19:26.29 | *** join/#asterisk diatonic (~diatonic_@mail.clearwater-research.com) |
| 19:26.49 | cusco_ | how does that relate to asterisk? |
| 19:27.06 | Aut0ExeC | I think i may have fixed in cdr.conf... gmttime=yes |
| 19:28.11 | Aut0ExeC | do I have to restart asterisk when I make a change to cdr? |
| 19:28.21 | \DSAFEW\ | ferdna, is it registered? |
| 19:28.22 | ferdna | cusco_, it doesn't... |
| 19:28.45 | ferdna | \DSAFEW\, yes... it registers normally... and can receive calls |
| 19:28.48 | ferdna | but can't make |
| 19:29.28 | cusco_ | Aut0ExeC: try module reload cdr |
| 19:29.28 | \DSAFEW\ | hmmm, sorry, sounds like a provider issue |
| 19:29.43 | cusco_ | or config reload /etc/asterisk/cdr.conf |
| 19:29.47 | Aut0ExeC | cusco_: thanks |
| 19:30.33 | Aut0ExeC | last question... whats the best prepaid provider? |
| 19:30.41 | Aut0ExeC | no monthly charges |
| 19:30.49 | Aut0ExeC | with roll over |
| 19:32.28 | rogersja | Aut0ExeC: google voice ;) |
| 19:33.11 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-130-106.twcny.res.rr.com) |
| 19:34.37 | Aut0ExeC | rogersja: lol i'm using that now |
| 19:34.52 | jaytee | I'm getting one-way jitter on the outbound RTP for external calls all of a sudden. |
| 19:34.52 | Aut0ExeC | rogersja: looking for a backup plan for when google decides to stop allowing free to US |
| 19:35.06 | Besticles | [May 3 03:01:04] WARNING[26575]: channel.c:1044 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/apexoutbound@apexdial-9126;1 What exactly is this warning trying to tell me? Should I take it literally, or could it represent a deadlock is occuring? |
| 19:37.17 | Aut0ExeC | anyone here know how long google is allowing free to US? |
| 19:37.32 | *** join/#asterisk weinerk (~weinerk@unaffiliated/weinerk) |
| 19:37.37 | *** join/#asterisk titter (~Justin@c-98-208-153-148.hsd1.fl.comcast.net) |
| 19:38.34 | weinerk | How can I make a call announcement/confirmation from AGI script? |
| 19:38.35 | weinerk | So that receiving party can near announcement and/or confirm accepting call, |
| 19:38.35 | weinerk | meanwhile caller hears MOH/ringin/etc. |
| 19:38.38 | p3nguin | I think I heard it would end in 2012. |
| 19:38.57 | Aut0ExeC | du |
| 19:38.58 | Aut0ExeC | sux |
| 19:39.14 | p3nguin | Hmm, I thought du was pretty good at what it does. |
| 19:39.18 | Aut0ExeC | lol |
| 19:39.24 | Aut0ExeC | typo |
| 19:39.50 | Aut0ExeC | yeah i hope it continues to be free |
| 19:39.59 | Aut0ExeC | it was supposed to end this year |
| 19:40.04 | Aut0ExeC | but they extended it |
| 19:40.22 | rogersja | i bet google keeps it up |
| 19:40.32 | rogersja | seems like the sort of thing they'd do |
| 19:40.33 | Aut0ExeC | i hope so man |
| 19:40.55 | rogersja | not that I'd know though ;) |
| 19:41.02 | Aut0ExeC | lol |
| 19:41.26 | Aut0ExeC | they may just be tryin to get the market first |
| 19:41.30 | Aut0ExeC | get free exposure |
| 19:41.35 | Aut0ExeC | then start taggin on |
| 19:41.38 | Aut0ExeC | most companies do that |
| 19:41.54 | Aut0ExeC | then u decide to be loyal and stay |
| 19:42.02 | Aut0ExeC | and buy their package |
| 19:42.02 | rogersja | perhaps for ancillary services, but i think the core product is here to stay |
| 19:42.10 | Aut0ExeC | ok |
| 19:42.15 | rogersja | at the current rate :P |
| 19:43.07 | Aut0ExeC | lol |
| 19:43.45 | Aut0ExeC | only thing is you still need a real voip provider if u want a real number |
| 19:43.45 | Aut0ExeC | its only good for callin g |
| 19:43.51 | Aut0ExeC | but not getting calls |
| 19:44.01 | Aut0ExeC | so either way.. ur still gonna need a number |
| 19:44.14 | rogersja | you get a gv number when you sign up |
| 19:44.23 | rogersja | it is dialable from pstn |
| 19:44.30 | Aut0ExeC | free? |
| 19:44.39 | rogersja | i have a few of them in different area codes |
| 19:44.42 | rogersja | yes free |
| 19:44.44 | Aut0ExeC | no way |
| 19:44.56 | Aut0ExeC | i never noticed |
| 19:45.01 | rogersja | free US did with every account :P |
| 19:45.16 | rogersja | you didn't sign up with google voice yet did you? |
| 19:45.21 | rogersja | you're just using google talk |
| 19:45.36 | Aut0ExeC | ohh |
| 19:45.39 | Aut0ExeC | theres a diff |
| 19:45.40 | gruvfunk | good stuff |
| 19:45.47 | Aut0ExeC | ok |
| 19:46.30 | Aut0ExeC | yeah i'm using google voice tho.. where to see my number? |
| 19:46.45 | rogersja | top right |
| 19:47.34 | gruvfunk | I have some GV accounts where signing up for a DID was optional - you may have to look for that option |
| 19:47.51 | Aut0ExeC | ok |
| 19:49.05 | Aut0ExeC | perhaps because i'm not in the US |
| 19:49.05 | rogersja | the only thing is i do have issues setting the CID on outgoing calls :P |
| 19:49.18 | Aut0ExeC | ya |
| 19:49.37 | rogersja | Aut0Exec: yes you do need to do it from a us ip ;) |
| 19:50.36 | gruvfunk | it's free for users in the US |
| 19:50.42 | Aut0ExeC | lol oh ok |
| 19:50.48 | Aut0ExeC | well that explains it |
| 19:50.51 | gruvfunk | proxy yourself |
| 19:50.58 | rogersja | its free for users in canada too, if you signup with a us IP address |
| 19:51.05 | gruvfunk | :) |
| 19:51.25 | Aut0ExeC | looks likes i'm gonna neex a proxy :P |
| 19:51.28 | Aut0ExeC | *need |
| 19:52.20 | gruvfunk | Aut0ExeC, hit me up if you need |
| 19:52.31 | weinerk | Anybody? Please help re: call announce/confirm from AGI script |
| 19:55.44 | _Corey_ | weinerk: You could accomplish what you want easily with a Dial() argument |
| 19:55.45 | ferdna | by googling around that error is: Unsupported Media Type |
| 19:55.51 | ferdna | what does this means? |
| 19:55.58 | _Corey_ | weinerk: core show application dial |
| 19:57.26 | weinerk | thanks _Corey_ I see for example here: |
| 19:57.27 | weinerk | http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 19:57.27 | weinerk | option M(x): Executes the macro (x) upon connect of the call (i.e. when the called party answers). |
| 19:57.27 | weinerk | or |
| 19:57.27 | weinerk | option A(x): Play an announcement (x.gsm) to the called party. |
| 19:57.27 | weinerk | so I am close... but not yet |
| 19:57.57 | ferdna | ok i see... ipcop modifies the packets or something? |
| 19:58.03 | ferdna | anyone has any experience with this? |
| 19:58.15 | *** join/#asterisk mac-mini (~mac-mini@unaffiliated/macmini/x-648924) |
| 19:59.11 | _Corey_ | weinerk: M() is what you want |
| 19:59.22 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 19:59.42 | _Corey_ | Play your sound and ask the callee to accept in your macro |
| 19:59.48 | _Corey_ | Set(MACRO_RESULT=CONTINUE) if yes |
| 19:59.56 | _Corey_ | Set(MACRO_RESULT=ABORT) if not |
| 20:02.50 | weinerk | _Corey_ , (1) thanks again (2) does the macro have to be in a etxconf dialplan or can I somehow do it all from AGI? |
| 20:03.24 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 20:03.42 | *** join/#asterisk luckman212 (~irc@pool-74-108-1-53.nycmny.fios.verizon.net) |
| 20:07.23 | _Corey_ | weinerk: AGI != dialplan, so the macro must be in the dialplan |
| 20:07.37 | _Corey_ | you can run the Dial() command from the AGI with that parameter if you wantr |
| 20:12.30 | *** join/#asterisk movozo (~Mo@p57AA0203.dip0.t-ipconnect.de) |
| 20:17.28 | weinerk | _Corey_: thanks, that is what I am trying - |
| 20:17.31 | weinerk | something like |
| 20:17.33 | weinerk | Dial("Local/14083334444#@from-internal|20|M(confirm^custom/prompt^^123)m(default)t|14083334444# |
| 20:17.47 | weinerk | ? |
| 20:18.34 | _Corey_ | Why the number again at the end? |
| 20:19.39 | weinerk | sorry |
| 20:19.43 | weinerk | got that wrong |
| 20:20.23 | *** part/#asterisk rogersja (~RogersJa@S0106000f6695039f.gv.shawcable.net) |
| 20:21.50 | _Corey_ | Otherwise it looks like you have the idea |
| 20:21.59 | _Corey_ | best to experiment and see how it works |
| 20:26.22 | *** join/#asterisk Polysics (~Luca@host51-72-dynamic.41-79-r.retail.telecomitalia.it) |
| 20:26.24 | Polysics | hello |
| 20:26.48 | Polysics | chances someone knows why my freshly installed asterisk isn't picking up configs in /etc/asterisk? |
| 20:27.11 | pabelanger | Polysics: $ sudo make samples |
| 20:27.12 | Polysics | 1.8 installed with configure/make/checkinstall with just picking the mysql addons |
| 20:28.02 | Polysics | pabelanger, they are in /etc/asterisk now |
| 20:28.17 | Polysics | yet i created an extension and it isn't showing in the dialplan in the console |
| 20:28.18 | pabelanger | Polysics: of course :p |
| 20:28.42 | pabelanger | what does your asterisk.conf look like? |
| 20:28.53 | pabelanger | did you change the ASTETCDIR var? |
| 20:29.24 | Freeaqingme_ | Polysics, did you restart asterisk after changing the config files? |
| 20:29.55 | Polysics | restarting now - btw, does 1.8 not have "dialplan reload"? |
| 20:29.59 | weinerk | _Corey_: can I send you a small thankyou? |
| 20:30.06 | gruvfunk | Polysics: yes |
| 20:30.11 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-130-106.twcny.res.rr.com) |
| 20:30.14 | _Corey_ | haha, no worries |
| 20:30.18 | Freeaqingme_ | Polysics, it does |
| 20:30.32 | weinerk | seriously - you spent time on me |
| 20:30.47 | gruvfunk | share and share alike |
| 20:31.06 | *** join/#asterisk lost_sou1 (~noymfb@cpe-67-249-130-106.twcny.res.rr.com) |
| 20:31.51 | *** join/#asterisk rogersja (~RogersJa@S0106000f6695039f.gv.shawcable.net) |
| 20:33.34 | _Corey_ | just help someone else out when the time comes |
| 20:34.53 | Polysics | where do all these extensions come from now? |
| 20:35.08 | Polysics | i would like a minimal SIP config, just running some tests |
| 20:35.19 | Polysics | there's something like an ael_demo clogging up the dialplan |
| 20:35.45 | cusco_ | clean extensions.ael file |
| 20:36.13 | weinerk | _Corey_: I will try, but seriously would love to paypal you at list a little thanks |
| 20:38.21 | keith4 | someone needs to come up with a remote-buy-a-beer service |
| 20:38.31 | _Corey_ | lol |
| 20:38.33 | keith4 | like, some sort of beer gift certificate, good at any bar |
| 20:38.46 | Polysics | i suppose i botched something in the sip config |
| 20:38.52 | Polysics | is bindaddr mandatory? |
| 20:38.57 | _Corey_ | weinerk: I'll be at Astricon, you can buy me a beer during the open bar ;) |
| 20:39.16 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 20:40.12 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 20:40.21 | p3nguin | If you leave out bindaddr, I believe it will default to 0.0.0.0, which is all addresses on all interfaces. |
| 20:40.52 | weinerk | :-) |
| 20:41.34 | Polysics | ok, clients do register, i am not hearing the "hello world" though |
| 20:41.41 | Polysics | let me pastie some confs |
| 20:42.24 | *** join/#asterisk cerberus_za (~coert@196-215-29-217.dynamic.isadsl.co.za) |
| 20:44.24 | Polysics | http://pastie.org/1861819 |
| 20:44.29 | Polysics | relevant configs |
| 20:44.43 | Polysics | i can call 9, asterisk correctly reports the call, but i do not hear the audio |
| 20:44.53 | Polysics | no errors, client is behind a NAT, might it be that? |
| 20:45.57 | gruvfunk | Polysics: there are 3 "dialplan" configuration files provided in the samples: extensions.conf, extensions.lua and extensions.ael. If you are going to use the .conf, I find it best to either delete or rename the lua and ael configs. |
| 20:46.13 | Polysics | gruvfunk, i did exactly that |
| 20:46.17 | gruvfunk | k |
| 20:46.32 | Polysics | i also renamed the .conf to only have the few i wanted for the mooment |
| 20:46.52 | gruvfunk | NAT |
| 20:47.10 | sxpert | is evil |
| 20:47.18 | gruvfunk | if your client is behind it, you want a nat=yes |
| 20:47.30 | gruvfunk | sxpert: agree |
| 20:48.01 | Polysics | nat is evil, but nat=yes solved my problem :-D |
| 20:48.06 | Polysics | thanks to all |
| 20:48.20 | gruvfunk | necessary evil |
| 20:49.12 | sxpert | useless evil. switch to ipv6 ;) |
| 20:50.05 | cj | :) |
| 20:50.22 | cj | sxpert: IPv6: It *mostly* works! |
| 20:50.42 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-130-106.twcny.res.rr.com) |
| 20:51.12 | cj | speaking of which, I need to set up the IPv6 filters on the colo firewall... |
| 20:52.38 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 20:53.41 | *** join/#asterisk De_Mon (de_mon@fl-71-49-14-47.dhcp.embarqhsd.net) |
| 20:54.07 | carrar | ipv6 is a passing fad |
| 20:54.25 | Polysics | well, it does look like a solution to a lot of things |
| 20:55.05 | De_Mon | We got some SPA501G's a few days ago and I got the configured and they register but I can't get a dialtone... |
| 20:55.11 | carrar | Once they commercialize ipv4 addresses for resale without the need for ARIN, it's a free market! |
| 20:55.47 | carrar | IPv6 is fun to toy around with |
| 20:56.11 | carrar | been using it since 2001 |
| 20:56.33 | gruvfunk | De_Mon: no dialtone at all? did you buy used devices that are possibly bad? |
| 20:58.02 | sxpert | carrar: a free market that will fail |
| 20:58.14 | carrar | haha |
| 20:58.16 | carrar | hardly |
| 20:58.28 | carrar | it's already started |
| 20:58.36 | sxpert | carrar: thing is, ip addresses don't "belong" to people, they are "lent" by arin |
| 20:58.41 | De_Mon | gruvfunk I can push the setup button and get audio, but no dial tone |
| 20:58.44 | carrar | with MSFT purchase of IP's addresses from Nortel |
| 20:58.52 | sxpert | carrar: whoever thinks otherise is a fool |
| 20:58.57 | carrar | yeah you hear that over and over |
| 20:59.03 | carrar | yes |
| 20:59.04 | carrar | yet |
| 20:59.05 | sxpert | carrar: stupid judge doesn't understand squat |
| 20:59.15 | carrar | people are still buying and sellign IP addresses |
| 20:59.49 | sxpert | carrar: which doesn't mean they aren't fools |
| 21:00.06 | sxpert | they just get ripped off by the thieves "selling" the addresses |
| 21:00.43 | carrar | ARIN doesn't have any authority to stop anything btw |
| 21:01.18 | sxpert | carrar: the US are lost anyhow |
| 21:01.35 | sxpert | so the problem is moot |
| 21:01.37 | sxpert | ;) |
| 21:01.57 | carrar | the US started the internet |
| 21:02.03 | leifmadsen | I started the internet |
| 21:02.04 | carrar | you're welcome |
| 21:02.08 | sxpert | besides, just by lending away zillions of ipv6 blocks, ipv4 will go away |
| 21:02.08 | leifmadsen | don't let Al Gore fool you |
| 21:02.14 | carrar | heh |
| 21:02.25 | carrar | Are you my tube!! |
| 21:02.26 | gruvfunk | lol |
| 21:02.38 | _Corey_ | It's all tubes you know |
| 21:02.39 | sxpert | and all those expensively bought ipv4 addresses will later be worth squat |
| 21:03.10 | carrar | thats the tech market for ya |
| 21:03.22 | carrar | everything is worth squat at some point |
| 21:03.41 | leifmadsen | invested heavily in tulips |
| 21:03.46 | carrar | yeah |
| 21:03.53 | carrar | I was just at hte tulip festical |
| 21:03.55 | carrar | festival |
| 21:04.09 | carrar | http://pics.osburn.com/photo/47769/original |
| 21:04.33 | carrar | no shortage of interest there |
| 21:04.35 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:04.52 | carrar | http://pics.osburn.com/photo/47812/original |
| 21:04.57 | gruvfunk | carrar: Alb? |
| 21:05.11 | carrar | WA |
| 21:05.13 | sxpert | doesn't give a shit though... has a /21 and a /22 :) |
| 21:06.59 | gruvfunk | is hungry |
| 21:11.10 | *** join/#asterisk andyoutside (6161c2a5@gateway/web/freenode/ip.97.97.194.165) |
| 21:16.02 | andyoutside | Ok I have a server I just upgraded asterisk on and it now crashes every so often. I can go in and do asterisk -r and interact with it but it does not show calls. |
| 21:19.03 | \DSAFEW\ | andyoutside, latest version? what's the build info? |
| 21:20.56 | andyoutside | Asterisk 1.8.3.3 built by root @ localhost on a i686 running Linux on 2011-05-02 05:04:21 UTC |
| 21:21.41 | jkroon | andyoutside, using SIP over TCP? |
| 21:21.46 | andyoutside | on centos 1.6 |
| 21:22.14 | andyoutside | Yes to SIP would have to check if over tcp |
| 21:22.55 | jkroon | chainsaw reported a bug that kills asterisk under certain conditions when doing sip over tcp. can't remember the details. |
| 21:23.13 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 21:23.41 | neurosys | Whow! Was just testing an * box and asterisk just crashed and reran itself. Anyone seen anything like this? |
| 21:25.05 | andyoutside | er that is CentOS Linux 5.6 |
| 21:25.39 | jkroon | neurosys, asterisk_safed |
| 21:25.56 | neurosys | oh nm. It appears they are running out of ram |
| 21:25.58 | andyoutside | does it completely kill asterisk or just part of it? |
| 21:25.58 | jkroon | there are numerous variants of it, but asterisk has to crash crash for those to kick in. |
| 21:26.29 | jkroon | andyoutside, iirc it segfaults. |
| 21:27.46 | andyoutside | Segfaults? I do not know what you mean by that. |
| 21:36.52 | jkroon | andyoutside, then it most likely doesn't apply to you. |
| 22:04.08 | *** join/#asterisk Marvelous (~CE0@197.195.75.32) |
| 22:09.58 | *** join/#asterisk ftoad (~rmiloh@70-36-143-28.dsl.dynamic.sonic.net) |
| 22:10.42 | *** join/#asterisk rogersja (~RogersJa@S0106001346faf79f.gv.shawcable.net) |
| 22:17.42 | *** join/#asterisk chazzam (~chazz@173-24-239-247.client.mchsi.com) |
| 22:18.48 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 22:19.41 | *** join/#asterisk cesar_CR (~cesar@201.193.82.8) |
| 22:30.46 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-130-106.twcny.res.rr.com) |
| 22:34.06 | *** join/#asterisk [netman] (~netman@20.Red-80-39-52.staticIP.rima-tde.net) |
| 22:39.38 | *** join/#asterisk cyford (~cyford@96-25-169-243.gar.clearwire-wmx.net) |
| 22:40.50 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-130-106.twcny.res.rr.com) |
| 22:46.07 | andyoutside | Ok I have a server I just updated asterisk to 1.8.3.3 from 1.8.0 on centos 5.6 and it now crashes every so often. I have asterisk -r up and it looks like it stoped mid call routing. It still responds to commands. But core restart does not work. |
| 22:46.57 | andyoutside | I use crash lossly more it stops working as it should. |
| 22:50.49 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 22:51.46 | carrar | mmm |
| 22:51.48 | carrar | crashes |
| 22:52.23 | jkroon | tzafrir_laptop, are you around by any chance? |
| 22:52.33 | tzafrir_laptop | jkroon, yes |
| 22:52.39 | *** join/#asterisk remnant13 (~Gray@unaffiliated/remnant13) |
| 22:52.46 | *** join/#asterisk km2 (~km2@99-117-98-49.lightspeed.sntcca.sbcglobal.net) |
| 22:52.53 | jkroon | tzafrir_laptop, i know you are somewhat familiar with a lot of the BRI code in dahdi-linux. |
| 22:53.16 | jkroon | i'm having an issue where zlen ends up being < 3 on one span in particular. |
| 22:53.25 | jkroon | how do I go about debugging it? |
| 22:53.37 | tzafrir_laptop | what do you mean by "zlen"? |
| 22:54.05 | jkroon | well, sruffel made the original commit: http://www.mail-archive.com/svn-commits@lists.digium.com/msg31958.html |
| 22:54.27 | jkroon | but basically I'm seeing this in dmesg: wcb4xxp 0000:04:00.0: odd, zlen less then 3? |
| 22:54.48 | jkroon | http://pastebin.com/mz7xwrgD for the full trace. |
| 22:56.23 | jkroon | from what I can see it looks like the drivers is only obtaining two bytes from the hardware and it requires at least three. |
| 22:56.30 | km2 | experiencing an odd issue on asterisk 1.6.2.16.1 where upon answering an incoming call, ringing is heard by the callee (but not caller). this is very intermittent and not reproducible. where could i start troubleshooting? |
| 22:57.50 | jkroon | km2, channel types involved would be a good start. |
| 22:58.58 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 22:59.29 | km2 | this only happens on calls from DAHDI to SIP channels |
| 23:00.25 | jkroon | analog? isdn? bri? pri? |
| 23:00.30 | km2 | oh, PRI |
| 23:00.57 | km2 | just found this (exact issue), reading up now: http://forums.asterisk.org/viewtopic.php?f=1&t=76458 |
| 23:01.19 | jkroon | Your Dial() string? |
| 23:01.33 | km2 | except i'm using polycom 550s |
| 23:02.30 | km2 | looking up Dial() string |
| 23:02.57 | km2 | would this have anything to do with echo cancellation by any chance? |
| 23:05.46 | *** join/#asterisk imox1234 (~imox1234@p4FC5C507.dip0.t-ipconnect.de) |
| 23:06.35 | km2 | this is the log line from the last call that had this issue: http://pastebin.com/7F06uawM |
| 23:07.04 | jkroon | km2, what EC are you using? |
| 23:07.27 | jkroon | Dial() looks fine. |
| 23:07.57 | jkroon | (i've seen some really messed up stuff with echotraining=yes and oslec) |
| 23:08.42 | km2 | the HWEC on the sangoma a101 |
| 23:08.59 | km2 | i'll check echotraining and so forth |
| 23:20.36 | carrar | *YAWN* |
| 23:21.01 | carrar | cj, you made the list |
| 23:21.05 | carrar | You're famous now |
| 23:21.33 | carrar | http://www.seattleix.net/docs/20110428_Annual_Meeting_Minutes.html |
| 23:22.29 | tzafrir_laptop | jkroon, sorry for the delay. Not really sure |
| 23:23.36 | tzafrir_laptop | jkroon, echotraining=yes should be harmless with any recent version of OSLEC |
| 23:23.43 | tzafrir_laptop | but it's pointless anyway |
| 23:30.46 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-130-106.twcny.res.rr.com) |
| 23:32.11 | *** join/#asterisk jrrose (~Jon@207.98.203.74) |
| 23:32.35 | andyoutside | would 1.8.3.3 use more memory than 1.8.0? |
| 23:34.02 | jrrose | I'd imagine. |
| 23:34.41 | p3nguin | I don't see any way to determine that based entirely on the version numbers. |
| 23:35.04 | leifmadsen | p3nguin: :) |
| 23:35.06 | *** join/#asterisk mclaro (~mclaro@190.244.79.220) |
| 23:35.07 | jrrose | Whenever a new configuration option is added for instance, you'll have new stuff that needs stored. |
| 23:35.24 | andyoutside | and I am talking RAM |
| 23:35.26 | leifmadsen | jrrose: but no new options would be added to a branch |
| 23:35.28 | Freeaqingme | and new optimizations may have been put in |
| 23:35.36 | Freeaqingme | andyoutside, just give it a shot on a testbox |
| 23:35.41 | p3nguin | No new configuration options should have been created between version 1.8.0 and version 1.8.3. |
| 23:35.43 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 23:35.43 | leifmadsen | I'm curious how you came to the conclusion/theory that it would use more memory |
| 23:35.46 | Freeaqingme | it really does depend on all circumstances involved |
| 23:36.05 | leifmadsen | throws out a big round maybe |
| 23:36.20 | jrrose | I'm just saying I don't think it would be especially abnormal for something to change that makes it bigger. |
| 23:36.35 | jrrose | And given the natural direction of things to become more complex over time... |
| 23:36.36 | andyoutside | well right now on 1.8.3.3 it is using 75% |
| 23:36.43 | andyoutside | I do not recall how much it used before |
| 23:36.45 | p3nguin | Does it use more RAM? Possibly. Does it use more RAM because the version number is greater? Absolutely not. |
| 23:36.50 | andyoutside | but I am having problems |
| 23:36.59 | p3nguin | 75% of what? |
| 23:37.10 | jrrose | Of course it isn't because the version number is greater... |
| 23:37.12 | Freeaqingme | of 100? :P |
| 23:37.22 | p3nguin | He has 100M RAM? I doubt that. |
| 23:37.25 | leifmadsen | 75% of infinity |
| 23:37.27 | Freeaqingme | 100% |
| 23:37.32 | andyoutside | 512 |
| 23:37.49 | p3nguin | 75% of a total of 512M RAM is a lot for Asterisk. |
| 23:38.12 | jrrose | I'm used to seeing it stated in terms of an amount of RAM rather than as a percentage. |
| 23:38.30 | p3nguin | I don't even have 384M RAM in total... if Asterisk used that much, we'd have a problem. |
| 23:38.31 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:38.55 | jrrose | enable the memory debugging compilation flag if you want to be able to spit out a specific amount that Asterisk is using. |
| 23:39.12 | Freeaqingme | p3nguin, it does once again depend on a lot of factors. If the bloke runs a million channels simultaneously I'd say it's low |
| 23:39.43 | leifmadsen | the more modules you have loaded, the more memory you'll use as well |
| 23:39.52 | p3nguin | Until we see the entire picture, I have to assume that's not utilizing any channels for calls at all. |
| 23:39.53 | leifmadsen | try unloaded some modules you're not using and see what happens |
| 23:40.56 | andyoutside | 159312 kB/usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c |
| 23:41.30 | andyoutside | so something is being bad and might not be asterisk |
| 23:42.18 | jrrose | Even then it's still nowhere near 75% of 512MB |
| 23:43.30 | p3nguin | 155.6M RAM isn't that bad. |
| 23:44.19 | p3nguin | That's only around 30%. |
| 23:45.05 | andyoutside | the 75 was total in use |
| 23:46.03 | andyoutside | ok after reboot total at 50% |
| 23:46.18 | p3nguin | How are you determining total in use? |
| 23:46.51 | carrar | easy, higher version number == higher amount of RAM!! :) |
| 23:47.01 | p3nguin | haha |
| 23:47.47 | sxpert | soooo true |
| 23:48.25 | andyoutside | letting freepbx report it to me |
| 23:49.19 | p3nguin | I have no idea what FreePBX is reporting because I don't use FreePBX. If you want to see what your system is using, login via ssh and run top and/or free to determine actual RAM usage. |
| 23:51.01 | andyoutside | Mem: 514376k total, 488288k used, 26088k free, 21276k buffers |
| 23:51.02 | ruben23 | hi i have done - make menuselect on asterisk 1.6 and i found thie on selecttion this means..? --- XXX chan_h323 ---> this mean its not included..? or not loaded..? |
| 23:52.56 | p3nguin | I'd personally rather see the "-/+ buffers/cache" line of `free' to say what your actual memory usage is. |
| 23:53.19 | p3nguin | Of course you can read it and figure it out by yourself... it's in plain text. |
| 23:56.29 | carrar | mmm text |
| 23:59.10 | *** join/#asterisk neurosys (~neurosys@c-65-34-188-197.hsd1.fl.comcast.net) |
