| 00:00.03 | Dovid | ChannelZ: I tried to look at that. that was way over my head. any "simpler" way to get a variable and use it in the php script? |
| 00:00.22 | Dovid | i wonder if it is my version of asterisk. it works with other scripts of mine |
| 00:00.28 | Dovid | on othe serers |
| 00:01.27 | ChannelZ | I dunno I just do it myself (opening stdin/stdout and writing commands/reading responses), I haven't used any of the wrappers. |
| 00:05.06 | Dovid | ChannelZ: mind to PB some sample code? |
| 00:06.52 | ChannelZ | http://pastebin.com/YvDP4SCx |
| 00:07.17 | ChannelZ | That doesn't really parse anything but it shows you very simply how to talk to AGI |
| 00:08.10 | Dovid | ChannelZ: thanks. gona test it |
| 00:08.13 | ChannelZ | you could use preg_match or whatever to yank out the results in whatever ways make sense for the response to the command |
| 00:09.07 | ChannelZ | (it's dumping what it receives from Asterisk using error_log if you have PHP logging to a file somewhere..) |
| 00:13.39 | *** join/#asterisk cyborg-one (~cyborg-on@212-178-22-148.broadband.tenet.odessa.ua) |
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| 00:48.54 | mostym | hey guys, I'm getting UNREACHABLE when I do sip show peers for a majority of my extensions. They stay OK for about 5-10 minutes then enter the UNREACHABLE state. |
| 00:48.58 | mostym | Any ideas? |
| 00:50.03 | mostym | THe firewall has beeen disabled. |
| 00:52.55 | WIMPy | Are they behind NAT? |
| 00:53.17 | mostym | yes. |
| 00:53.50 | WIMPy | Then it's almost certainly their firewall forgetting the "connection". |
| 00:54.06 | mostym | how do i fix? |
| 00:54.38 | WIMPy | Configure the natting FW, enable keppalives if available or lower the qualifyfreq. |
| 00:55.54 | mostym | keep alive is enabled on the voip phones to 30 sec. |
| 00:56.06 | mostym | should I decrease that? |
| 00:56.26 | WIMPy | If you can;t increase the timout on the FW, yes. |
| 00:57.15 | mostym | any idea where that would be on the fw? |
| 00:57.38 | WIMPy | That obviousely depends on the type of FW. |
| 00:58.40 | mostym | They have a RV042G router. |
| 00:58.42 | WIMPy | It could have run out of memory as well. |
| 01:03.52 | mostym | WIMPy, anyway to find that out on the Rv042G router? |
| 01:06.56 | *** join/#asterisk Nemus (~Nemus@c-67-161-251-38.hsd1.ut.comcast.net) |
| 01:07.38 | WIMPy | I don't use Cisco for more than switching. |
| 01:08.22 | mostym | can you recommend a good smb router/firewall combo? |
| 01:08.52 | WIMPy | I prefer to use linux. |
| 01:10.02 | mostym | appliance? |
| 01:10.14 | WIMPy | Whatever. |
| 01:10.43 | WIMPy | But a full kernel is a plus. |
| 01:11.06 | mostym | gotcha. |
| 01:22.14 | lvlinux | linux or m0n0wall or pfsense |
| 01:22.44 | lvlinux | people rave about m0n0wall for appliances (mostly embedded) |
| 01:23.31 | lvlinux | depends on your needs really |
| 01:25.33 | mostym | I used to use one called smoothwall or something. |
| 01:25.38 | mostym | like 5 years ago. |
| 01:31.08 | mostym | WIMPy, decreasing the keepalive time to 15 sec seems to be working. |
| 01:31.17 | mostym | should I be concerned about memory issues? |
| 01:32.09 | *** join/#asterisk deo (~dnepangue@112.198.79.88) |
| 01:32.31 | *** join/#asterisk nightrid3r (~kvirc@94-227-155-32.access.telenet.be) |
| 01:37.59 | lvlinux | i wouldn't worry about it if you are not taxing the router with other things - i.e. lots of VPN sessions and such. |
| 01:39.48 | WIMPy | mostym: You need to ask the FW if it has some some shortage. |
| 01:40.46 | lvlinux | does that router run Cisco IOS or is it something else? |
| 01:41.21 | lvlinux | if it runs IOS you can from the console run "show processes" and it will show you the load and memry free and such |
| 01:47.23 | *** join/#asterisk Naikrovek (~wwhite@unaffiliated/naikrovek) |
| 01:48.35 | mostym | lvlinux, it's a smb router. |
| 01:51.13 | lvlinux | yes i know, but i didn't know if it ran ios or not. I guess not though - should still have a status page that you can check and it might say about the memory. |
| 01:53.20 | lvlinux | is going to eat a fantastic homemade meal :-) |
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| 05:57.05 | kontinuity | hi folks |
| 05:57.25 | kontinuity | do most of the IP phones support SRTP and TLS-SIP? |
| 05:57.30 | kontinuity | specs really don't tell much |
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| 06:41.05 | gg608f | hi all |
| 06:41.13 | gg608f | do you guys know if Asterisk 10.9.0 have Web sockets support? |
| 06:41.23 | gg608f | http show status dont show /ws |
| 06:44.42 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 06:45.05 | kaldemar | gg608f: no. |
| 06:45.48 | gg608f | thanks |
| 06:45.57 | gg608f | will upgrade to * 11 |
| 06:45.58 | gg608f | thanks |
| 06:46.28 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
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| 07:35.25 | *** join/#asterisk Emmanuel18 (AcidBurn@per18-2-88-165-49-175.fbx.proxad.net) |
| 07:35.34 | Emmanuel18 | Bonjour |
| 07:35.52 | Emmanuel18 | Apres avoir remis iptables en route je ne peux plus passer d'appels ni en recevoir |
| 07:36.02 | Emmanuel18 | j'ai ouvert les port 10000:20000 et 5060 |
| 07:36.08 | Emmanuel18 | mais j'ai un message d'erreur |
| 07:36.22 | Emmanuel18 | WARNING[21009]: chan_sip.c:3425 __sip_xmit: sip_xmit of 0x1dbc9e0 (len 400) to 212.27.52.5:5060 returned -1: Operation not permitted |
| 07:36.22 | Emmanuel18 | [Oct 12 09:30:44] NOTICE[21009]: chan_sip.c:13294 sip_reg_timeout: -- Registration for '0952966838@freephonie.net' timed out, trying again (Attempt #2) |
| 07:36.36 | wdoekes | did you block OUTPUT? |
| 07:37.15 | Emmanuel18 | no |
| 07:37.17 | wdoekes | and you did set -p udp, right? |
| 07:38.08 | Emmanuel18 | I want you to put my iptables? |
| 07:38.26 | *** join/#asterisk darkskiez (~mhb@fsf/member/darkskiez) |
| 07:38.27 | wdoekes | ~pb |
| 07:38.27 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 07:39.20 | Emmanuel18 | http://pastebin.com/t7TCiuUE |
| 07:39.40 | Emmanuel18 | this is my iptable |
| 07:40.36 | wdoekes | iptables -t filter -P OUTPUT DROP |
| 07:41.04 | wdoekes | iptables -t filter -A OUTPUT -p tcp --dport 5060 -j ACCEPT |
| 07:41.10 | Rac-on | there is a 'iptables -t filter -A OUTPUT -p tcp --dport 5060 -j ACCEPT', but you probably want to include UDP too here |
| 07:41.22 | wdoekes | 09:36 < wdoekes> did you block OUTPUT? |
| 07:41.22 | wdoekes | 09:37 < Emmanuel18> no |
| 07:41.23 | wdoekes | 09:37 < wdoekes> and you did set -p udp, right? |
| 07:41.44 | Rac-on | bows to wdoekes. |
| 07:42.31 | Emmanuel18 | I am a long answer because I used google translate to answer I am French. |
| 07:43.03 | wdoekes | remove the "iptables -t filter -P OUTPUT DROP" |
| 07:43.14 | wdoekes | use "iptables -t filter -P OUTPUT ACCEPT" |
| 07:43.25 | wdoekes | and remove all -A OUTPUT |
| 07:43.41 | Emmanuel18 | -A INPUT ! |
| 07:45.27 | Emmanuel18 | I delete « iptables-t filter-P OUTPUT DROP » and-A OUTPUT? |
| 07:45.47 | wdoekes | correct |
| 07:46.01 | wdoekes | but first you add: iptables -t filter -P OUTPUT ACCEPT |
| 07:46.59 | wdoekes | there is really no reason to block output, unless you're really paranoid about the apps on your machine, but than your firewall would look a lot differently anyway |
| 07:47.04 | wdoekes | s/than/then/ |
| 07:49.06 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 07:49.49 | Emmanuel18 | It's good like that |
| 07:49.56 | Emmanuel18 | http://pastebin.com/18XNE87v |
| 07:49.58 | *** join/#asterisk timahvo1 (~rogue@197.180.204.104) |
| 07:51.59 | wdoekes | line 11 is a no-op. move that to line 1 |
| 07:52.18 | wdoekes | #!application only works on the first line |
| 07:52.52 | wdoekes | and you need to add this: iptables -t filter -P OUTPUT ACCEPT |
| 07:53.15 | wdoekes | if it's DROP now, and you run this firewall script, you'll lose all output |
| 07:53.57 | wdoekes | you haven't removed all other lines with OUTPO |
| 07:54.01 | wdoekes | *OUTPUT yet |
| 07:54.25 | Emmanuel18 | I remove iptables-t filter-P OUTPUT ACCEPT |
| 07:54.47 | wdoekes | + iptables -t filter-P OUTPUT ACCEPT |
| 07:54.58 | wdoekes | - iptables -t filter -A OUTPUT* |
| 07:55.20 | Emmanuel18 | The OUTPUT # ASTERISK or everywhere? |
| 07:56.00 | wdoekes | ajouter: iptables -t filter -P OUTPUT ACCEPT |
| 07:56.07 | wdoekes | éliminer tout -A OUTPUT |
| 07:56.24 | Emmanuel18 | ok |
| 07:58.38 | Emmanuel18 | It seems to work there is more error messages. |
| 07:59.21 | Emmanuel18 | thank you |
| 07:59.47 | wdoekes | pas du problème |
| 07:59.59 | Emmanuel18 | ;) |
| 08:00.11 | Emmanuel18 | pas « de » problemes |
| 08:00.12 | Emmanuel18 | ;) |
| 08:00.20 | wdoekes | excusez moi |
| 08:00.28 | Emmanuel18 | no soucis |
| 08:00.30 | wdoekes | (probably wrong too) |
| 08:00.30 | *** join/#asterisk deo (~dnepangue@122.53.72.218) |
| 08:00.55 | Emmanuel18 | Is that the installation is complicated voicemail? |
| 08:01.55 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 08:01.56 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 08:10.58 | wdoekes | Emmanuel18: no, try it.. a matter of setting up voicemail accounts in voicemail.conf, calling VoiceMail() in your dialplan for leaving voicemail and VoiceMailMain() to read voicemail. |
| 08:12.18 | Emmanuel18 | wdoekes you do outsourcing |
| 08:16.01 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:16.02 | wdoekes | Emmanuel18: wrong question. but if you have work, I have a boss you can contact on info at osso.nl |
| 08:16.42 | Emmanuel18 | ok |
| 08:17.27 | Emmanuel18 | http://pastebin.com/5XG1hz9w |
| 08:18.42 | wdoekes | you need [freephonie-out] in your sip.conf |
| 08:19.00 | wdoekes | (or in your /etc/hosts, but that's probably not what you mean) |
| 08:28.51 | *** join/#asterisk k610 (~Instantbi@cred.epid.ucl.ac.be) |
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| 08:39.41 | *** join/#asterisk Maliuta (nikolai@donetsk.lusan.id.au) |
| 08:41.17 | funky1 | hi all :) I was wondering if someone could help me with some tips, i have a 3g usb modem huawei e153 to be specific, i have unlocked it and enabled the voice function on it, i put in the sim card of my phone, now i want to get it working with asterisk, haveing some trouble though, i have downloaded the chan_dongle module and installed, but what i don't get is this, does asterisk make the connection to my provide by itself or do i need to estab |
| 08:41.18 | funky1 | lish a connection somehow outside of asterisk? |
| 08:45.26 | Naikrovek | the connection should be established when you power the module up, i assume. |
| 08:45.44 | Naikrovek | an easier way to do what you want is to use your cellphone but use chan_bluetooth to get asterisk to talk to it |
| 08:45.58 | Naikrovek | if that's still around, anyway, that chan_bluetooth |
| 08:46.41 | Naikrovek | you'd put a USB bluetooth dongle in yoru computer if you dont already have a BT radio then pair it with your phone, then tell asterisk to use it |
| 08:47.27 | Naikrovek | it can be set up so that incoming cell calls go straight to asterisk, and outgoing voip calls leave through your cell phone (though they won't be voip when they leave your phone, just regular cell calls) |
| 08:51.15 | funky1 | hm haven't thought about doing it that way, will have a go at it, thank you for the tip :) |
| 08:51.30 | *** join/#asterisk alexturner (~admin@onebig.lnk.telstra.net) |
| 08:51.47 | kaldemar | so can chan_dongle. you just don't have to deal with an icky bluetooth interface in between. |
| 08:53.34 | alexturner | Hey guys, i've got a question thats making me pull my hair out. Currently I have a FreePBX/asterisk install running on a VM and moving it over to a Elastix box - we have a FXO->SIP unit that sends inbound PSTN calls straight to the IP of the FreePBX server. |
| 08:54.11 | fling | kaldemar: hey :p what is wrong with this? -> exten => _[5,9]XXXXXXX,1,Goto(749${EXTEN},1) |
| 08:54.23 | kaldemar | fling: nothing. |
| 08:54.31 | alexturner | I've moved the config files from the freepbx server over to the elastic box, though the elastix box does not respond to the incoming sip calls from the FXO->SIP unit |
| 08:54.40 | fling | kaldemar: I want it to match every 8-digit number starting with 5 or 9 |
| 08:54.52 | kaldemar | fling: there might be something wrong with your expectations of what it does. |
| 08:55.07 | fling | yes |
| 08:55.11 | alexturner | Other than IP tables, why won't an inbound call even appear in the asterisk console (verbosity 10)? |
| 08:55.24 | kaldemar | fling: and that's what your pattern should do. |
| 08:55.34 | alexturner | I'm getting to the stage where i feel like throwing my feces around the room |
| 08:55.43 | kaldemar | fling: no, wait. [5,9] should be [59]. |
| 08:57.36 | kaldemar | alexturner: are you using SIP? is the call sent somewhere else? disable any blocking of incoming traffic in iptables and use tcpdump to verify that you are getting incoming traffic. also, enable sip debug with "sip set debug on". |
| 08:57.43 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 09:00.21 | alexturner | okkk |
| 09:00.22 | alexturner | working on it |
| 09:00.35 | alexturner | nothing in iptables |
| 09:02.02 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 09:02.47 | fling | kaldemar: right, now it works, thanks |
| 09:03.09 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 09:05.06 | alexturner | kaldemar: agh! It's appearing in sip debug though the auth is bad - do incoming calls have to auth? how do they? are they an extension? |
| 09:07.29 | kaldemar | alexturner: whether they authenticate or not is up to you to configure. "are they an extension?" <-- does not make sense |
| 09:08.15 | alexturner | yes i understand, an no |
| 09:08.34 | alexturner | basically its 8 fox lines -> queue |
| 09:08.37 | alexturner | *FXO |
| 09:11.03 | kaldemar | for configuring the SIP side in asterisk, ask in #freepbx or #elastix. |
| 09:28.17 | *** join/#asterisk deo (~dnepangue@203.177.214.75) |
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| 09:40.13 | *** join/#asterisk Ecco (~User@82.241.237.172) |
| 09:40.31 | Ecco | Hi everyone. I have a working asterisk setup (pretty simple, just using SIP). |
| 09:40.43 | Ecco | Sound quality is horrible though. How can I diagnose what's going on? |
| 09:52.12 | mirela666 | Hello, i'm having a wierd issue with SayPhonetic and digit 9, it's totaly ignoring it, is it bacause 9_p.gsm file (which I deleted) |
| 09:52.21 | mirela666 | I guess it's expecting it there |
| 09:54.52 | bulkorok | Ecco: mak a pcap trace and inspect in wireshark => VoIP |
| 09:55.19 | Ecco | well, what will that tell me really? Thing is, it "works". |
| 09:55.54 | Ecco | It's just a problem of quality: people say they have trouble hearing |
| 09:56.55 | bulkorok | what people say and who the quality really is, is a big difference usually... |
| 09:57.17 | bulkorok | try tracing |
| 10:01.55 | Ecco | ok, I'll give it a try |
| 10:02.06 | Ecco | But to be honest I don't really know where/what to look at |
| 10:02.28 | Ecco | I've done protocol inspection before, but more in a "boolean" fashion (figuring out why it doesn't work at all) |
| 10:03.02 | Ecco | it's not clear how I'll be able to "see" poor audio quality in wireshark |
| 10:10.24 | *** join/#asterisk sdr_ (~sdr@teracomm-group.com) |
| 10:11.28 | sdr_ | hi all, |
| 10:13.17 | sdr_ | i've got this lovely patch set for 1.6.2 about automated filtering of SIP attackers and I wander will it be worthy to port it to current asterisk |
| 10:13.29 | sdr_ | is there a chance this will go mainstream? |
| 10:15.20 | sdr_ | I also have a patches about easy routing of calls when behind multi gate NAT ... |
| 10:15.22 | bulkorok | sdr_: fail2ban is doing that... |
| 10:16.34 | *** join/#asterisk alexturner (~admin@203-173-37-59.perm.iinet.net.au) |
| 10:16.36 | bulkorok | Ecco: just check the "telephony" menu in wireshark... there is RTP as option. take a look at "all streams" |
| 10:16.42 | sdr_ | bulkorok, fail2ban is something not couple of hundred lines and involves much more resources .. |
| 10:16.51 | Ecco | ok |
| 10:16.54 | Ecco | I'll give it a try |
| 10:17.17 | bulkorok | sdr_ : Btw is 1.6 EOL !? |
| 10:17.37 | bulkorok | ah... see your question... |
| 10:17.41 | sdr_ | yes it is i'm thinking porting it to 10.x |
| 10:17.58 | bulkorok | ask in #asterisk-dev |
| 10:18.28 | sdr_ | :) |
| 10:21.35 | *** join/#asterisk Kalamatee (~ChatPhobi@82.132.136.148) |
| 10:22.23 | *** join/#asterisk alexturner (~admin@onebig.lnk.telstra.net) |
| 10:26.23 | funky1 | hm still working with my 3g usb modem, got a little bit further, but what is still unclear to me is if i need to connect to the network of the SIM outside from asterisk with e.g. network manager or do i have to configure the network provider settings somewhere in asterisk? how do i asterisk to connect to the network, it shows my dongle but jus tsays not connected |
| 10:29.36 | zamba | _X.404 will match all extensions that end with '404', right? |
| 10:35.51 | kaldemar | zamba: all extensions that start with 0-9, are at least 5 characters long and end with 404. |
| 10:38.45 | kaldemar | funky1: afaik you need to configure dongle.conf and then asterisk will take care of it. |
| 10:40.01 | funky1 | kaldemar: there are no settings where i can set the network provider in dongle.conf when i do "dongle show devices" in asterisk cli i see the dongle but it says "not connected" |
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| 10:46.10 | zamba | kaldemar: basically i want it to match on either 11 or 3 characters, which ends in 404.. how would that look? |
| 10:46.24 | kaldemar | zamba: characters or digits? |
| 10:46.40 | zamba | kaldemar: digits |
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| 10:47.08 | kaldemar | _XXXXXXXX404 and 404 as separate extensions. |
| 10:47.22 | zamba | oh |
| 10:47.33 | zamba | what about ends in 404 and MAY contain some digits before? |
| 10:47.41 | zamba | doesn't need to be, but can be |
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| 10:48.19 | kaldemar | there is no way to express such in patterns. |
| 10:51.03 | zamba | hm, ok.. |
| 10:51.31 | bulkorok | maybe with cut and goto/gotoif in dialplan |
| 10:51.47 | zamba | another question.. i want to set up a radio jukebox service by using shoutcast/icecast streams.. i've already set up one by using musiconhold, but i want to specify several different radio streams |
| 10:51.49 | zamba | is this possible? |
| 10:52.24 | bulkorok | you can strip teh exten with ${EXTEN:-3} afair to get the last three digits... |
| 10:52.47 | bulkorok | and then check with gotoif to send it to another exten |
| 10:53.24 | bulkorok | https://wiki.asterisk.org/wiki/display/AST/Manipulating+Variables+Basics |
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| 11:04.18 | zamba | hm, ok.. i'll take a look at that :) |
| 11:04.25 | zamba | what about the radio jukebox question? |
| 11:05.14 | *** join/#asterisk plantseeker (~Plantseek@77.240.63.97) |
| 11:05.35 | alexturner | hey all :) how can i turn auth off on inbound calls? (getting this one SIP/2.0 403 Forbidden (Bad auth)) |
| 11:09.51 | Tim_Toady | alexturner: dont define a password |
| 11:10.03 | alexturner | well it shouldn |
| 11:10.39 | kaldemar | alexturner: configure insecure=port,invite on the matching peer that has a static host configuration with host=<ip_address>. |
| 11:10.40 | alexturner | it shouldn't authenticate - it's a 3rd party FXO->SIP gateway thats forwarding inbound calls to the asterisk unit |
| 11:10.54 | alexturner | kaldemar: hello again :) |
| 11:11.27 | kaldemar | the fact that is is a gateway is no reason for it to not authenticate itself. |
| 11:13.06 | alexturner | Hmm, i'm just quite lost here |
| 11:13.19 | alexturner | Now I'm getting a SIP/2.0 401 Unauthorized |
| 11:13.25 | alexturner | so it seems as if the auth is happening |
| 11:13.55 | kaldemar | enable sip debug and pastebin the output of a call. |
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| 11:15.08 | alexturner | http://pastebin.com/H8PMXEJs |
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| 11:20.54 | kaldemar | alexturner: that's a registration attempt, not a call. |
| 11:21.02 | kaldemar | alexturner: what do you have in sip.conf? |
| 11:21.11 | alexturner | currently |
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| 11:21.48 | alexturner | nothing -_- |
| 11:22.27 | kaldemar | stuff won't just magically work by itself. you actually need to configure asterisk to use it. |
| 11:24.31 | alexturner | Yeah i understand, though on my last config I just set allow incoming calls from unauthenticated or something |
| 11:24.39 | alexturner | and she worked a charm with an inbound route |
| 11:25.22 | alexturner | [200] type=user nat=never secret=obs123 insecure=very dtmfmode=rfc2833 context=from-pstn authuser=200 |
| 11:27.08 | plantseeker | .news |
| 11:27.44 | alexturner | "Allow Anonymous Inbound SIP Calls?" is the option i was talking about before |
| 11:28.24 | kaldemar | that means nothing in asterisk. turn to #freepbx or #elastix with the configuration part. |
| 11:28.36 | alexturner | Yes, i know ;/ |
| 11:29.00 | kaldemar | also, insecure=very is invalid nowadays. i told you to use insecure=port,invite earlier. |
| 11:29.35 | kaldemar | and your device seems to think it is 207, not 200. |
| 11:29.55 | alexturner | thanks for looking mate |
| 11:29.59 | alexturner | looking into it now |
| 11:32.00 | Kalamatee | Hello, can anyone advise me where I might find a download for AsteriskNOW version 1.6 (I dont seem to be able to find it on the site)? |
| 11:34.35 | alexturner | to looks like a call now http://pastebin.com/mXvRBsD7 though still no good |
| 11:34.41 | alexturner | and no idea where the 201 is coming from |
| 11:36.06 | kaldemar | Kalamatee: http://www.asterisk.org/asterisknow |
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| 11:38.32 | kaldemar | alexturner: 200 came from the gateway configs. start by changing what you see in the From header (From: ""<sip:unknown@192.168.17.8>;tag=52417462a3270ce1) to something that can be identified by asterisk. next step is to handle "No matching peer for 'unknown' from '192.168.17.200:5062'". |
| 11:44.06 | *** join/#asterisk lminiero (~lminiero@host18-157-dynamic.18-79-r.retail.telecomitalia.it) |
| 11:44.28 | lminiero | hello everybody |
| 11:44.38 | lminiero | can anybody help me with a problem I've faced? |
| 11:44.50 | lminiero | I've written a Opus codec implementation for Asterisk 1.8 |
| 11:44.54 | lminiero | and it works fine |
| 11:45.05 | lminiero | but as soon as an audio hook is attached on a channel |
| 11:45.09 | lminiero | audio stops working |
| 11:45.44 | lminiero | is there anything the audio hooks interface expects that may be breaking in my case? |
| 11:46.03 | bulkorok | lminiero: maybe #asterisk-dev is a better pplace for you?! |
| 11:46.18 | lminiero | whoops sorry, didn't know about that other room... |
| 11:46.27 | bulkorok | np :-) |
| 11:46.32 | lminiero | thanks :) |
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| 12:24.42 | funky1 | is there any way i can further debug what is going wrong with my 3g usb modem i only get this in cli |
| 12:24.43 | funky1 | [dongle0] Error initializing Dongle |
| 12:24.43 | funky1 | <PROTECTED> |
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| 12:24.56 | funky1 | and google search does not bring up useful things |
| 12:25.09 | bulkorok | core debug |
| 12:25.11 | bulkorok | verbose |
| 12:25.28 | WIMPy | +set |
| 12:26.23 | funky1 | how do i enter that in cli? |
| 12:27.18 | WIMPy | core set verbose 9 |
| 12:27.30 | WIMPy | core set debug 9 |
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| 12:29.23 | funky1 | done that but does not show anything extra still same output |
| 12:30.02 | WIMPy | Does the Stick work otherwise? |
| 12:31.30 | funky1 | yes, it works on ubuntu without a problem and on my pbx system (centos) i can communicate and connect via minicom |
| 12:35.25 | WIMPy | Are you sure it is voice enabled? |
| 12:36.12 | funky1 | yep 100% checked it with dc unlocker and appropriate at command |
| 12:36.29 | funky1 | and on windows i can make calls with it |
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| 13:18.08 | Katty | morning |
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| 13:19.50 | eqbal | Hi , anyone figured out how to install asterisk with mountain lion ? |
| 13:20.30 | blitzrage | step 1) download linux step 2) format over mountain lion |
| 13:20.32 | blitzrage | :) |
| 13:20.54 | blitzrage | not a lot of development is done on osx, but it is possible. You just need to install the dependencies |
| 13:21.10 | eqbal | I keep getting this error when make menuselct menuselect_gtk.c:4:21: error: gtk/gtk.h: No such file or directory |
| 13:21.16 | eqbal | i have gtk tho |
| 13:21.36 | eqbal | ~/Downloads/certified-asterisk-1.8.11-cert7% brew install gtk+ Error: gtk+-2.24.11 already installed |
| 13:22.32 | eqbal | anyone successfully installed it ? i wanna use it with development @blitzrage with adhearsion , thanks for ur welling to help |
| 13:22.50 | *** join/#asterisk alexturner_ (~admin@203-173-37-59.perm.iinet.net.au) |
| 13:23.13 | blitzrage | eqbal: sounds like you don't have the development libraries for gtk though |
| 13:23.32 | blitzrage | beyond that, I can't really help as I use linux on my macbook pro |
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| 13:29.13 | Katty | blitzrage: hipster. |
| 13:29.20 | Katty | blitzrage: you were usin the linuxses before mac used them |
| 13:31.28 | carrar | *YA*W*N* |
| 13:31.48 | Katty | stuffs breakfast in carrar's yawn. |
| 13:32.03 | carrar | IcanHASespresso |
| 13:32.18 | carrar | must make |
| 13:32.37 | carrar | nomm nomm |
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| 13:50.32 | *** join/#asterisk fukuda_ (~IceChat9@183.181.70.86.rev.sfr.net) |
| 13:50.34 | fukuda_ | hi |
| 13:51.12 | newtonr | fukuda_: hello |
| 13:51.20 | *** join/#asterisk rjvvliet (~rjvvliet@smtp.cvanderlinden.nl) |
| 13:51.40 | fukuda_ | i'm a problem with asterisk and iaxmodem |
| 13:52.47 | fukuda_ | When I do: iax2 show peers, my modem alternates between the correct port is a random port. |
| 13:54.03 | fukuda_ | my project is the Fax to mail |
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| 14:08.57 | bulkorok | fukuda: why do you use iaxmodem!? |
| 14:11.46 | [TK]D-Fender | bulkorok, Why do you think? |
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| 14:15.46 | bulkorok | that's why I ask... I have no idea... |
| 14:16.33 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 14:17.08 | [TK]D-Fender | bulkorok, well "!?" implies some sort of shock ... which you should not have if yuo looked at what it's usually used with. |
| 14:17.50 | bulkorok | well-... usually with hylafax, but currently asterisk ReceiveFax is not that unstable |
| 14:17.58 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 14:18.00 | bulkorok | or sendfax |
| 14:18.26 | [TK]D-Fender | bulkorok, And will those apps integrate with desktop clietns, manager distribution lists, etc? |
| 14:18.57 | bulkorok | they could... |
| 14:19.16 | [TK]D-Fender | You mean ... if you invent an entire framework around them..... |
| 14:19.25 | [TK]D-Fender | Like ... reinventing Hylafax.... |
| 14:30.08 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 14:35.35 | carrar | !!!! |
| 14:35.38 | *** join/#asterisk kriger (~norge@68-190-90-80.dhcp.mdsn.wi.charter.com) |
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| 14:42.14 | zamba | [TK]D-Fender: to follow up yesterdays discussion about line keys and registrations.. we have the situation that someone has an active call and then wants to include a third party into this conversation by bridging all parties together to a conference.. is this possible with only one registration? or is this not related to that at all? |
| 14:42.29 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
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| 14:44.16 | [TK]D-Fender | zamba, It is not limited by registrations. It could be across multiple or not. The phone does the work. You need to support multiple calls. How you spread that over your line-keys is up to you. |
| 14:44.23 | WIMPy | zamba: With VOIP that is usually done in the phone. |
| 14:44.26 | *** join/#asterisk zpertee (~chatzilla@h130.205.17.98.static.ip.windstream.net) |
| 14:44.54 | [TK]D-Fender | zamba, You could use 1 line-key for 1 registration and support 8 calls on it, leaving 5 for BLF/speedial/etc |
| 14:45.28 | [TK]D-Fender | zamba, Or 1-call-per-key and have 2-3 linek-eys taken up by that 1 reg and haev a more natural view of your individual calls. |
| 14:46.23 | zamba | how do i know if i support multiple calls? |
| 14:46.35 | WIMPy | "RTFM" |
| 14:46.37 | zamba | i guess that's something i restrict/allow on the asterisk side? |
| 14:46.41 | *** join/#asterisk j4m3s_ (~j4m3s_@pdpc/supporter/active/j4m3s) |
| 14:46.49 | WIMPy | Nope. Phone. |
| 14:47.00 | zamba | oh, ok |
| 14:47.44 | WIMPy | You can restrict it with Asterisk, off course, but you have to actively do so. |
| 14:47.55 | zamba | ok.. that's good |
| 14:48.17 | [TK]D-Fender | zamba,You allocate X number of line-keys to your registration and Y number of calls per line-key. This is your PHONE setup. |
| 14:48.30 | [TK]D-Fender | zamba, X * Y = # of calls you can shuffle at a time |
| 14:48.55 | televoip | Does anyone have experience using Exchange VM with 1.8? I'm pulling my hair out trying to get REDIRECTING to work correctly. |
| 14:49.01 | [TK]D-Fender | zamba, your 650's can handle 24 <- How many you limit them to is up to you and how you want them to use their phones |
| 14:55.47 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:55.48 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 15:43.11 | zamba | [TK]D-Fender: ah, ok |
| 15:43.22 | zamba | i'll look into the settings on the phone for that.. thanks :) |
| 15:43.57 | zamba | but back to my previous question today.. i wanted to set up a radio jukebox of some kind.. when users dial to a certain extension they should get prompted for a number, where each number represent a different shoutcast/icecast stream |
| 15:44.00 | zamba | is this doable? |
| 15:44.30 | zamba | right now i have one stream running as the default moh |
| 15:45.47 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 15:49.17 | *** join/#asterisk navaismo (~navaismo@189.191.10.146) |
| 15:49.17 | [TK]D-Fender | zamba, then run multiple streams for multiple MoH |
| 15:50.08 | zamba | just one stanza per stream? |
| 15:52.14 | [TK]D-Fender | What does the config say? |
| 16:00.39 | *** join/#asterisk Arrnas (Arrnas@2002:5ee8:78be::5ee8:78be) |
| 16:02.45 | Arrnas | hey geys, i'm trying to make a call directly through a gsm gateway via openphone... i get g.729 packets with it and the gateway connects fine but i have no audio, audio settings look fine and i get the same results from my netbook |
| 16:04.50 | Arrnas | guys* |
| 16:04.52 | Arrnas | my bad |
| 16:13.05 | Qwell | wat |
| 16:13.17 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 16:20.14 | AkkerKid | Arrnas: you have a paid and working g729 codec on both ends? |
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| 16:35.43 | [TK]D-Fender | AkkerKid, 2 things : first, he's apparently not using Asterisk at all in there. Second... he's GONE |
| 16:36.23 | AkkerKid | Lunchtime! |
| 16:37.53 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
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| 16:56.03 | wonderworld | hey guys |
| 16:56.07 | cmendes0101 | hola |
| 16:56.09 | *** part/#asterisk hurdman (~ygcheny@bender.r0b0t.fr) |
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| 17:22.09 | *** join/#asterisk iulhk (iulhk@119.154.95.189) |
| 17:27.08 | iulhk | hello, anybody has any idea about this error, i am getting at my asterisk console " == Using SIP RTP CoS mark 5, == Using SIP VRTP CoS mark 6" ? |
| 17:27.23 | Qwell | What makes you think those are errors? |
| 17:31.05 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 17:34.44 | iulhk | <@Qwell>: i am dialing one of my DID from US, if i set cli format "LOCAL" all things work fine but just one problem, getting caller id without country code, if i set cli format E164, i got fast busy at caller-end who dialing this DID and got these lines at my asterisk cli "== Using SIP RTP CoS mark 5 |
| 17:34.44 | iulhk | <PROTECTED> |
| 17:34.44 | iulhk | <PROTECTED> |
| 17:34.44 | iulhk | <PROTECTED> |
| 17:34.44 | iulhk | "??? |
| 17:35.44 | drmessano | Those are not errors |
| 17:35.50 | drmessano | That has nothing to do with your call |
| 17:35.55 | drmessano | Pastebin the complete call |
| 17:35.59 | drmessano | ~pb |
| 17:36.00 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:37.08 | iulhk | <drmessano>: this is the main problem not getting any error at my asterisk console except these lines ;( |
| 17:42.00 | navaismo | turn on the sip debug and then pb |
| 17:42.55 | *** join/#asterisk lvlinux (~n1gg@c-50-147-64-9.hsd1.tn.comcast.net) |
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| 18:06.29 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 18:07.39 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 18:25.04 | *** join/#asterisk Sidrov (~Sidrov@AC812E5A.ipt.aol.com) |
| 18:25.08 | Sidrov | hello all |
| 18:25.57 | Sidrov | any asterisk geek here can tell me why asterisk 1.8.11 delete my call file from /var/spool/asterisk/outgoing/ BEFORE call hanging up ? |
| 18:26.04 | Sidrov | is it normal behave ? |
| 18:26.13 | Sidrov | i thought it's deteled only after hangup |
| 18:28.54 | kaldemar | the files are deleted upon read. |
| 18:28.59 | kaldemar | normal behavior. |
| 18:29.43 | jpsharp | Are they deleted on read or deleted on the call being answered? |
| 18:31.16 | *** join/#asterisk cusco_ (~tralala@2001:41d0:1:6caf::1) |
| 18:32.46 | Sidrov | after call is answered, after 5-10 seconds |
| 18:32.52 | Sidrov | but call is not hangup |
| 18:33.10 | Sidrov | is there any setting for that ? |
| 18:34.35 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 18:36.52 | navaismo | there are two options: |
| 18:36.53 | navaismo | AlwaysDelete: Yes/No - If the file's modification time is in the future, the call file will not be deleted |
| 18:36.54 | navaismo | Archive: Yes/No - Move to subdir "outgoing_done" with "Status: value", where value can be Completed, Expired or Failed. |
| 18:37.12 | Sidrov | found issue |
| 18:37.19 | navaismo | but not sure if that cover your needs |
| 18:37.34 | Sidrov | no, issue is about another asterisk -vvvgc running zombie |
| 18:38.20 | *** join/#asterisk kontinuity (~Adium@122.167.224.116) |
| 18:39.40 | Sidrov | the bloody zombie was deleting file |
| 18:58.52 | drmessano | Shoot it in the head |
| 19:00.07 | jpsharp | Nuke it from orbit |
| 19:02.44 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 19:03.46 | [TK]D-Fender | Nova its star |
| 19:05.07 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 19:05.17 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 19:05.30 | jmetro | well that escalated quickly |
| 19:07.44 | *** join/#asterisk aross42 (~aross@99.224.40.85) |
| 19:16.26 | jmetro | this google TISP connection is so fast |
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| 19:22.48 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 20:06.35 | *** join/#asterisk InHisName (~InHisName@96.25.37.57) |
| 20:08.43 | InHisName | I just switched ISPs. My asterisk and router are linux based. Asterisk is outside of NAT. I've updated the IP address changes. Everything but asterisk works. I cannot get any registrations to occur. Any clues or questions of me to help derive answer ? |
| 20:09.52 | Qwell | What does SIP debug say? Have you determined that your new ISP doesn't block SIP? |
| 20:10.11 | Qwell | (hint: Clearwire does) |
| 20:10.25 | jpsharp | They do? |
| 20:10.27 | jpsharp | makes note. |
| 20:10.30 | Qwell | They did. |
| 20:13.50 | InHisName | It was ok for a while, but I made a change to the 4G modem. I switched to passthru. Now eth1 shows external address (96.x.y.z) and several other things work better. I am not receiving any response to the registration inquiries. SIP debug just shows my packets going out(I assume). The rest of home network is functioning great. |
| 20:14.37 | InHisName | I can ping the IP from asterisk box for each site to register. |
| 20:14.56 | [TK]D-Fender | InHisName, You're still not showing SIP DEBUG for your failures. Stop asking us to advise you blind. |
| 20:15.38 | InHisName | You want the one way stream of packets trying to regsister in a pastebin ? |
| 20:16.43 | kontinuity | do all hard phones support SRTP and TLS-SIP? can you recommend a few good ones |
| 20:16.57 | kontinuity | I have my PBX on a remote server hence need the security |
| 20:17.13 | Qwell | kontinuity: No, they don't all support it. |
| 20:18.05 | kontinuity | Qwell: are there any good secure phones that you can recommend? |
| 20:19.34 | [TK]D-Fender | InHisName, Asterisk SIP debug |
| 20:20.34 | [TK]D-Fender | And don't go masking anything in there........ |
| 20:22.11 | rjvvliet | <PROTECTED> |
| 20:23.29 | InHisName | [TK]D-Fender: I may have a clue: before your message, I tried sip debug ip incoming.cheapvoip.com and I got the help message. Using the numeric address worked. Looks like it is having issues looking up addresses. |
| 20:23.42 | InHisName | asterisk 1.2.10 |
| 20:24.05 | [TK]D-Fender | that is just sad.... |
| 20:24.11 | InHisName | I changed dnsactual.conf to reflect new IP |
| 20:24.25 | InHisName | But still behaves same way. |
| 20:24.40 | [TK]D-Fender | And I guess we should expect to be in the dark on this the whole way through..... reclassifying now as "monologue" |
| 20:25.57 | [TK]D-Fender | checks out for the day |
| 20:26.01 | [TK]D-Fender | BBIAB |
| 20:27.43 | Qwell | ~upgrade asterisk |
| 20:27.43 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 20:30.43 | *** part/#asterisk keycruncher (~Adium@c-174-59-233-118.hsd1.pa.comcast.net) |
| 20:31.04 | InHisName | For anyone else following this: http://pastebin.com/Nr0ckMt5 here is the sip debug of registration attempts. |
| 20:31.56 | Qwell | InHisName: Upgrade. You're using 1.2. Nobody cares. |
| 20:42.00 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 20:42.00 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.9.0 (2012/10/08), 1.8.17.0 (2012/10/08), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.13 (2012/10/09) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 20:42.39 | *** join/#asterisk raub (~raub@ip70-171-42-89.ga.at.cox.net) |
| 20:43.37 | raub | What is the proper way to install Asterisk in a centos 5 box: http://www.asterisk.org/downloads/yum or https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages ? |
| 20:44.45 | Qwell | raub: The latter. |
| 20:45.17 | raub | Qwell: thanks! |
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| 21:36.30 | mathi | hi |
| 21:36.37 | WIMPy | lo |
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| 22:13.07 | *** join/#asterisk branderghost (~izaak@69.158.87.159) |
| 22:17.03 | branderghost | where can i read about different styles of writing/calling Gosub-style sub-routines? i.e. 1) merging sub-routine contexts and calling them via. labelled priorities, vs. 2) keeping the contexts separate and calling them via. respective context? |
| 22:17.21 | branderghost | i ask because the sample config with 1.8 takes the former approach, but i can't find any information as to why |
| 22:19.19 | *** join/#asterisk Neptu (~Neptu@c213-89-2-159.bredband.comhem.se) |
| 22:20.09 | [TK]D-Fender | Because the samples aren't always the best proofed and While it may work it also might not be what someone else considers a "best practice" |
| 22:24.24 | branderghost | ok. weird. guess i'll keep my sub-routines in separate contexts then as that seems to be the only approach i see documented. i really wonder why 1.8's sample config doesn't do that, though… it itches! |
| 22:28.50 | mjordan | branderghost: my preference - keep em separated out by extension if nothing else (context is a +1 IMO). Always make sure you have a Return() at the end. |
| 22:36.27 | branderghost | asterisk is so weird |
| 22:41.06 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-anwtkjmbthshuvlt) |
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| 23:01.38 | *** join/#asterisk alexturner (~admin@onebig.lnk.telstra.net) |
| 23:05.02 | *** part/#asterisk alexturner (~admin@onebig.lnk.telstra.net) |
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| 23:28.40 | *** join/#asterisk alexturner (~admin@onebig.lnk.telstra.net) |
| 23:28.57 | alexturner | Hey guyssss, how can i allow inbound sip calls to avoid this message "SIP/2.0 403 Forbidden (Bad auth)" |
| 23:30.29 | jpsharp | Get your username & passwords to match on both Asterisk and the SIP client. |
| 23:30.46 | pabelanger | alexturner, using an ARM processor? |
| 23:31.15 | alexturner | No, x86_64 |
| 23:32.00 | alexturner | I've tried everything, the bastard grandstream just won't authenticate |
| 23:33.08 | pabelanger | ~collectdebug |
| 23:33.08 | infobot | i heard collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 23:33.16 | pabelanger | alexturner, pb your SIP debug |
| 23:34.31 | alexturner | tada http://pastebin.com/dqJBZTNF |
| 23:34.54 | alexturner | i have set the granstream to authenticate as 200 |
| 23:34.58 | alexturner | though it insists on 201 |
| 23:35.00 | alexturner | no idea why |
| 23:35.11 | alexturner | i think it's because it's crap |
| 23:41.48 | *** join/#asterisk Rahoul (~nicolas@186.18.112.219) |
| 23:42.22 | alexturner | This is all on a secure internal network |
| 23:42.27 | alexturner | and isn't open to the outside |
| 23:42.48 | alexturner | how would I allow all inbound calls? |
| 23:51.53 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 23:52.31 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip66-104-72-222.z72-104-66.customer.algx.net) |
| 23:53.15 | pabelanger | alexturner, No matching peer for 'unknown' from '192.168.17.200:5062' |
| 23:53.38 | pabelanger | either add [unknown] as a sip peer or fix your user auth on the phone |