| 00:05.29 | *** join/#asterisk lopz (~jjlc@unaffiliated/lopz) |
| 00:05.38 | lopz | hi guys :) |
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| 00:58.45 | cneb3000- | hmm -- been a while since I've been here ^_^ |
| 00:58.46 | cneb3000- | ... |
| 00:58.48 | cneb3000- | lurks |
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| 04:20.23 | raden | is there any best distro for asterisk to run on ? |
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| 04:30.08 | schultza | why do i get a segmentation fault when i just change one letter (username) in sip.conf when restarting asterisk? |
| 04:35.20 | *** join/#asterisk kieppie (~jaco@ip-58-28-154-35.static-xdsl.xnet.co.nz) |
| 04:36.16 | kieppie | hi guys. where is the AsteriskNOW installation script for CentOS - to install the AsteriskNOW system (Asterisk & FreePBX) on a [clean] CentOS host, please? |
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| 04:37.20 | [TK]D-Fender | kieppie: there is none, because AsteriskNOW is a complete distro, not a "script" |
| 04:37.58 | kieppie | thanks. so I'd be looking for the FreePBX installer then? |
| 04:39.17 | [TK]D-Fender | There is no drop-in for RPM systems. There is a 3rd party script made for debuian based systems, but that's it |
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| 04:41.19 | kieppie | I think that's what I was thinking of, thanks. |
| 04:41.31 | ThinkGNU- | Hello everyone |
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| 05:07.50 | ThinkGNU- | I know you can setup if statements based on a time and use that for an after hours dialplan. How can I set it up so a user can over ride that if needed? |
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| 05:52.02 | qakhan | hi all |
| 05:52.47 | qakhan | how can i connect two asterisk server with each other while one has static IP and other has dynamic |
| 05:54.15 | ChannelZ | tell them host=dynamic or host=whatever.com as appropriate |
| 05:54.26 | ChannelZ | use dnsmgr |
| 05:55.01 | ChannelZ | If you're using dyndns of some sort anyway |
| 05:56.17 | mpe | and have the dynamic one do a register to the static one |
| 05:57.26 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-txqehwmhuiksguhq) |
| 06:01.13 | qakhan | but voice traffic would not be save |
| 06:01.19 | qakhan | right? |
| 06:01.29 | ChannelZ | eh? |
| 06:02.16 | ChannelZ | If it changes its IP in the middle of a call... not a lot you can do about that |
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| 06:10.04 | schmidts | good morning |
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| 06:23.21 | kleszcz | morning |
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| 06:28.13 | qakhan | i m getting this message while i m installing unimrcp /usr/bin/ld: cannot find -lpocketsphinx |
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| 06:48.07 | qakhan | i m getting this message while i m installing unimrcp /usr/bin/ld: cannot find -lpocketsphinx |
| 06:53.43 | wdoekes2 | install libpocketsphinx1 ? |
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| 07:04.06 | qakhan | wdoekes2 u there? |
| 07:04.26 | wdoekes2 | irrelevant |
| 07:04.49 | qakhan | k |
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| 07:18.57 | din3sh | good mrning all |
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| 07:43.49 | jacc0 | hi all!\ |
| 07:44.35 | jacc0 | I'm looking for people who have access to a cisco SPA phone to do some tests |
| 07:44.42 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 07:45.07 | jacc0 | I've found a security hole as big as a skyscraper |
| 07:45.14 | *** join/#asterisk cneb3000- (~cneb3000@5ac8d14e.bb.sky.com) |
| 07:45.21 | ChannelZ | I do but not physically in front of me |
| 07:45.46 | jacc0 | :) |
| 07:45.47 | jacc0 | nice |
| 07:46.03 | jacc0 | what type of Cisco SPA? |
| 07:46.45 | ChannelZ | actually.... they are Linksys, so depending on what series you're talking about they either have something or nothing in common :) |
| 07:46.51 | jacc0 | I want to see in what types of spa phone the security hole exists |
| 07:46.59 | ChannelZ | SPA942 |
| 07:47.32 | jacc0 | it needs to support XML push |
| 07:47.51 | jacc0 | that allows you to controle alle the phones features by pushing an XML to it |
| 07:48.21 | ChannelZ | well they are provisioned via XML but I'm not sure if I'd call it 'push' per se |
| 07:49.16 | jacc0 | https://supportforums.cisco.com/docs/DOC-9932 |
| 07:49.38 | jacc0 | lol you can push configuration to the phone without authentication |
| 07:49.40 | jacc0 | :P |
| 07:50.24 | jacc0 | I've reported this issue more then a year ago; the say the resolved it in the new cisco spa firmware; but they did not |
| 07:50.31 | jacc0 | I can still use any username and password |
| 07:50.44 | jacc0 | to push te XML |
| 07:51.27 | jacc0 | it even allows you to install new firmware :p |
| 07:52.30 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 07:52.32 | ChannelZ | I'd have to try when I'm in the office tomorrow |
| 07:53.05 | jacc0 | I have this Cisco spa remote controle |
| 07:53.30 | jacc0 | I've made it a couple of months ago |
| 07:53.51 | jacc0 | you can download it here: http://www.securitysource.eu/ |
| 07:53.55 | jacc0 | it's made for windows |
| 07:54.06 | jacc0 | it'/s just a POC |
| 07:54.27 | *** join/#asterisk Tim_Toady (~fuzzy@188.4.93.97.dsl.dyn.forthnet.gr) |
| 07:54.42 | jacc0 | all you need to do is enter the phones ip and you might be able to controle it remoteley :) |
| 07:54.44 | ChannelZ | Are you calling your own program a Piece Of Crap? |
| 07:54.52 | jacc0 | yes :) |
| 07:54.59 | jacc0 | Proof of consept |
| 07:55.05 | ChannelZ | Ah. |
| 07:55.07 | jacc0 | ;) |
| 07:55.22 | jacc0 | but it hangs when you enter a non-SPA ip |
| 07:55.30 | jacc0 | so it's not realy stable |
| 07:55.38 | jacc0 | but it does the job |
| 07:56.06 | ChannelZ | well mine aren't locked down at all so I'll have to configure a password and stuff and see. I'm assuming you've turned off provisioning and/or upgrading |
| 07:56.20 | jacc0 | could you check if it works on the phone you have? and report that back to me? |
| 07:56.36 | ChannelZ | yeah I'll let you know tomorrow |
| 07:56.54 | jacc0 | ' so I'll have to configure a password and stuff and see' : that is the problem, you can not configure a username password for the xml push in the old firmware |
| 07:57.38 | jacc0 | the new firmware allows you to confige en username and password but it doesn't respect it: you still can push XML using any username and password combination you like |
| 07:59.24 | ChannelZ | But with provisioning turned off it still fetches the config? |
| 07:59.43 | jacc0 | not sure, I will check that : thanks for the input |
| 08:00.22 | ChannelZ | The have an option "User configurable resync" too but I'm not sure what the distinction is. |
| 08:00.36 | jacc0 | (btw it is not only about pushing the config you can also send 'keypresses' for every key on the phone luike you are sitting behind it |
| 08:00.47 | jacc0 | :) |
| 08:01.21 | jacc0 | you can also send an xml to do a factory default rese |
| 08:01.25 | jacc0 | *reset |
| 08:01.46 | jacc0 | (it will clear username and password set for the web interface :) |
| 08:02.10 | din3sh | I got a SPA525G |
| 08:02.16 | jacc0 | :) me to |
| 08:02.20 | jacc0 | tested that one already |
| 08:02.33 | din3sh | looks neat |
| 08:02.39 | jacc0 | 'as big as a skyscraper' |
| 08:03.00 | jacc0 | if you have windows installed; check the link i just posted : http://www.securitysource.eu/ |
| 08:03.16 | jacc0 | and see if it works on your phone |
| 08:03.27 | jacc0 | you might want to download the latest firmware |
| 08:03.49 | din3sh | cisco 79xx series are much better |
| 08:04.11 | jacc0 | you can download it here : https://supportforums.cisco.com/docs/DOC-9923 |
| 08:04.59 | jacc0 | the release note states it resolves issue CSCts08768 'the phone does not respect the authentication when |
| 08:04.59 | jacc0 | pushing XML code' |
| 08:05.02 | jacc0 | but it does not |
| 08:05.18 | jacc0 | as fare as I've checked |
| 08:05.58 | jacc0 | I would love to know if the remotecontrole also works for the 79xx series |
| 08:08.09 | din3sh | remote control? |
| 08:08.17 | din3sh | http://www.securitysource.eu/<---------------this you mean? |
| 08:08.32 | din3sh | i have a bunch of 79xx right in front of me |
| 08:08.34 | din3sh | lol |
| 08:09.20 | jacc0 | :) yes that one |
| 08:09.23 | jacc0 | I've made it |
| 08:09.26 | din3sh | brrr requires new version of .net |
| 08:09.31 | din3sh | to install |
| 08:09.41 | jacc0 | you can also get the source if you would like to compile it your self |
| 08:09.57 | din3sh | its a provisional tool? |
| 08:10.04 | din3sh | provisioning* |
| 08:11.43 | jacc0 | let me lookup the featur I used (i'll post a link) |
| 08:12.00 | jacc0 | CiscoIPPhoneExecute |
| 08:12.23 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:12.24 | jacc0 | SPA 79xx also supports it :) |
| 08:12.37 | jacc0 | http://www.voip-info.org/wiki/view/Cisco+79XX+XML+Push |
| 08:13.13 | din3sh | you can push config files using your application ? |
| 08:13.22 | jacc0 | you can even push firmware :) |
| 08:13.33 | jacc0 | not with my application |
| 08:13.51 | jacc0 | my application allows you to press buttons like you are sitting behind the phone |
| 08:13.59 | jacc0 | :) |
| 08:14.49 | din3sh | in what scenario for example do you use the remotecontrol application? |
| 08:15.50 | jacc0 | It's just a Proof of Consept to shouw the security hole in the phone |
| 08:16.07 | jacc0 | I also added war-dialing functionality |
| 08:16.43 | jacc0 | and a button to factory default the phone (so it removes the username and password needed to login to the webinterface |
| 08:16.52 | jacc0 | :) |
| 08:17.13 | jacc0 | I've also made asterisk module |
| 08:17.16 | din3sh | you can do that without authentication? |
| 08:17.17 | din3sh | :o |
| 08:17.19 | jacc0 | yes |
| 08:17.23 | jacc0 | you can |
| 08:17.56 | din3sh | thats the security hole you wanna to uncover |
| 08:17.58 | din3sh | :o |
| 08:18.03 | jacc0 | yes |
| 08:18.07 | jacc0 | I did a year ago |
| 08:18.17 | jacc0 | and cisco says they resolved it |
| 08:18.21 | jacc0 | but they didn't |
| 08:18.27 | jacc0 | as far as I can check |
| 08:18.59 | din3sh | they are very slow with their support stuff |
| 08:19.02 | jacc0 | it realy is 'as big as a skyscraper' |
| 08:19.28 | jacc0 | they told me to buy a support contract if i wanted this issue to be resolved |
| 08:19.39 | jacc0 | they are crazy, stupid and blind |
| 08:20.14 | jacc0 | I'm not going to pay them for reporting a security hole |
| 08:20.35 | jacc0 | I guess they are from another planet |
| 08:22.44 | din3sh | have you tried the latest firmware 9.x on cisco 79xx series? |
| 08:23.03 | jacc0 | I've this reporter wanting to publish about it (again) |
| 08:23.24 | jacc0 | So I want to find out what types of cisco phones are affected |
| 08:23.51 | din3sh | am having to install .net framework to test it |
| 08:24.38 | jacc0 | you could also open port 80 to the internet and provide me with the ip. They I can give it a try :) |
| 08:25.22 | jacc0 | if you don't mind me controling your phone :) |
| 08:25.23 | *** join/#asterisk mpe (~mpe@109.70.54.56) |
| 08:26.52 | jacc0 | you can read the first article about it here : http://webwereld.nl/nieuws/109269/beveiliger-publiceert-cisco-telefoonhacktool.html\ |
| 08:27.14 | jacc0 | (you might want to use google translate because it is in dutch) |
| 08:28.29 | din3sh | real pain in the ass to get the port forwarding done |
| 08:28.31 | din3sh | loll |
| 08:29.13 | din3sh | ahh! linksys router acting b*tchy! |
| 08:29.29 | din3sh | they sell beta version products i guess |
| 08:30.16 | jacc0 | it is not; install teamviewer and SPI Port Forward |
| 08:30.30 | din3sh | its not i know |
| 08:30.32 | jacc0 | with teamviewer you can setup a reverce connect vpn |
| 08:30.40 | jacc0 | with your windows pc |
| 08:30.47 | din3sh | i have a router which freezes every couple of hours |
| 08:30.57 | jacc0 | with spi portforward you can forward port 80 to the phone :) |
| 08:31.26 | jacc0 | this way there is no need for port forwarding |
| 08:31.38 | jacc0 | http://vishnuvalentino.com/computer/spi-port-forward-redirection-for-windows-to-another-port-ip-address/ |
| 08:32.42 | *** join/#asterisk wonderworld (~ww@dsdf-4d0a1b23.pool.mediaWays.net) |
| 08:34.32 | din3sh | nice |
| 08:34.35 | din3sh | checking it out |
| 08:39.48 | *** join/#asterisk din3sh (~din3sh@41.212.202.82) |
| 08:40.50 | din3sh | jacc0: you from netherlands? |
| 08:40.57 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 08:41.01 | jacc0 | yes |
| 08:41.09 | din3sh | van persie land |
| 08:41.10 | din3sh | :p |
| 08:41.28 | jacc0 | hehehe |
| 08:41.36 | jacc0 | I'me that into soccer |
| 08:41.55 | din3sh | but big event coming up |
| 08:41.57 | din3sh | ;) |
| 08:42.22 | jacc0 | yes, and you would not say I'm not that into soccer if you would see me in my van :0 |
| 08:43.04 | jacc0 | http://nl.picturepush.com/public/4573952 |
| 08:43.09 | jacc0 | that's my van :) |
| 08:44.28 | din3sh | lolll |
| 08:44.37 | din3sh | already geared up |
| 08:44.44 | jacc0 | hehehe yes |
| 08:45.41 | din3sh | you might wanna change robben's pic though after last saturday |
| 08:45.43 | din3sh | :D |
| 08:48.26 | din3sh | kickass nonetheless |
| 08:48.27 | din3sh | ;) |
| 08:53.53 | jacc0 | hehehe I might |
| 08:53.54 | jacc0 | :P |
| 08:56.00 | cneb3000- | grrr voip carriers! |
| 08:56.06 | cneb3000- | Anyone know of a SIP trunk I can get set up like.. now.. and pay some credit |
| 08:56.22 | cneb3000- | I'm based in the UK - but international would be OK i guess |
| 08:56.43 | cneb3000- | only required for about £50 worth of calls |
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| 09:03.31 | chakrit | I couldn't access any page on wiki.asterisk.org atm. Is it down? |
| 09:07.54 | chakrit | oh, it's back up. |
| 09:11.28 | wdoekes2 | jacc0: I've tried pushing stuff to /CGI/Execute on a 504G.. here I seem to need the user username+password.. is there some magic you're doing to bypass the auth? |
| 09:14.47 | jacc0 | nope |
| 09:14.59 | jacc0 | what firmware version? |
| 09:15.15 | jacc0 | spa525g-7-5-1a ? |
| 09:15.17 | *** join/#asterisk s[x] (~sx]@ppp59-167-157-96.static.internode.on.net) |
| 09:15.25 | jacc0 | v7.5.1a ? |
| 09:15.42 | *** join/#asterisk joel_oliveira (~chatzilla@estrela-adm.nortenet.pt) |
| 09:15.47 | wdoekes2 | wrong phone I think :) |
| 09:17.37 | wdoekes2 | same software version though |
| 09:23.33 | din3sh | jacc0: The remote server returned an error (401) Unauthorized. |
| 09:23.47 | din3sh | cisco 7911, firmware 9.x.x |
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| 09:29.02 | jacc0 | okay |
| 09:29.07 | jacc0 | thanx for the test |
| 09:29.38 | din3sh | time out |
| 09:29.49 | din3sh | this time test with SPA508G |
| 09:30.01 | jacc0 | :) |
| 09:31.13 | din3sh | how about a tool to control/manage batch of spa or 79xx phones? |
| 09:31.14 | din3sh | ;) |
| 09:35.11 | jacc0 | I might look into that (if i can get my hands on a 79xx phone) |
| 09:36.16 | jacc0 | what is the resulton the SPA508G ? |
| 09:36.37 | jacc0 | and what firmware does it have? |
| 09:37.06 | din3sh | wait |
| 09:41.55 | din3sh | the phone not booting |
| 09:41.56 | din3sh | lolll |
| 09:46.32 | *** join/#asterisk chakrit_ (~chakrit@ec2-175-41-204-38.ap-northeast-1.compute.amazonaws.com) |
| 09:47.37 | jacc0 | it might need POE |
| 09:49.09 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
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| 09:55.05 | din3sh | using POE itself |
| 09:58.04 | din3sh | might be a problem with phone itself |
| 09:58.29 | din3sh | reset it to factory defaults, the screen is blank now |
| 09:58.30 | din3sh | :o |
| 10:06.34 | din3sh | jacc0: firmware version : 7.4.4 |
| 10:06.48 | din3sh | provision: yes upgrade:yes |
| 10:06.58 | din3sh | the application stops responding |
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| 10:16.11 | Goni | hi, my provider is asking me to send to different port than 5060,should I use host=xxx and port=xxx in the config? or host=xxx:port ? |
| 10:18.44 | kaldemar | host and port as separate parameters. |
| 10:18.59 | Goni | they are running in say, 5090 |
| 10:19.01 | kaldemar | you probably need to add the port to your register statement aswell. |
| 10:20.38 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 10:26.37 | din3sh | is there any advantage of using tcp instead of udp in *? |
| 10:28.08 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 10:29.27 | din3sh | cisco 79xx series firmware 9.x seems to work on tcp rather than udp |
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| 11:09.35 | jacc0 | din3sh: yes,for example when sending dtmf, when using udp and the network packets don't get to the other end, they won't be resend. With TCP they will be resend until ACK |
| 11:10.31 | kaldemar | only if SIP INFO is used to carry DTMF. |
| 11:11.30 | jacc0 | true |
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| 12:01.10 | bitwize | Hello everybody! I have an Asterisk 1.8 setup with a registred sip-trunk. I use call forwarding to external pstn-numbers with the Dial command for some incoming DIDs. My problem is that the forwarding-channel (created with the Dial-cmd) still remains open when the incoming call gets hanged up by the caller. How can I close the dialed channel when the caller hangs up? |
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| 12:13.17 | jacc0 | hangup() |
| 12:13.19 | jacc0 | ? |
| 12:15.34 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 12:16.01 | mirelab | bitwize: you can do Dial(,g) with that g option to continue in dialplan and then in next steps do Hangup() |
| 12:16.51 | mirelab | Dial(tech/ext/trunk,<timeout>,options) |
| 12:17.26 | bitwize | I'm using Hangup in the next stage after the Dial-command, but the call created with the Dial-cmd is still active. I will try the 'g' option, thanks! |
| 12:17.40 | mirelab | np :) |
| 12:17.55 | *** join/#asterisk din3sh (~din3sh@41.212.203.67) |
| 12:20.15 | kaldemar | g won't help, it continues to next priority when the destination channel hangs up. F might help. but it should really hang up anyway if asterisk detects the hangup from the caller. |
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| 12:29.37 | bitwize | kaldemar: Thanks! The F-option with a hangup-cmd works fine, I wonder why Asterisk wont detect this by itself... |
| 12:42.39 | *** part/#asterisk Bullmoose (~Bullmoose@71-37-168-220.bois.qwest.net) |
| 12:44.10 | bitwize | If i specify the option "canreinvite=no" on my peer (trunk) Asterisk makes a local bridging and detects the hangup as it should, but if Asterisk makes a remote bridging it wont close the called channel when the caller hangs up. |
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| 13:06.05 | Dovid | anyone here ever set up asterisk with a siemens PBX over a PRI/E1 ? |
| 13:06.48 | WIMPy | ~polls |
| 13:06.48 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
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| 13:07.00 | WIMPy | And yes, I have |
| 13:07.53 | Dovid | Wimpy: DO I need a cross over cable? |
| 13:08.20 | Dovid | when cable is not connected: |
| 13:08.20 | Dovid | May 22 16:04:03 pbx kernel: wanpipe1: Open Circuit is detected! |
| 13:08.21 | Dovid | May 22 16:04:03 pbx kernel: wanpipe1: Short Circuit is cleared!(2179) |
| 13:08.33 | Dovid | when it is all I get: wanpipe1: Short Circuit is cleared!(2179) |
| 13:12.24 | WIMPy | Depends on how the ports are wired. |
| 13:12.31 | WIMPy | Or if your card has jumpers. |
| 13:15.42 | WIMPy | You didn't even say which way you want to connect them. |
| 13:15.52 | danfromuk | Hello, ive just upgraded from 1.4 to 1.8 and i have an issue with the voicemail. If the call goes straight to the voicemail app it works fine. But if the call goes to a sip peer, times out and goes to the voicemail app, I hear alison as normal, then the beep, but the beep seems to continue and it doesnt record properly. |
| 13:16.19 | Dovid | WImpy: Which way meaning? i want to be master |
| 13:17.58 | danfromuk | It might be a codec issue. When it goes directly, asterisk plays gsm sound files. When it goes through the sip peer, asterisk plays ulaw alison sounds. |
| 13:18.17 | WIMPy | Dovid: Then you need a cross over cable, unless you can jumper your card. |
| 13:19.31 | Dovid | Wimpy: THanks. |
| 13:19.39 | Dovid | ok. I will create a cross |
| 13:20.28 | Dovid | http://help.fonality.com/TroubleshootingFAQ/Troubleshooting%28old%29/T1_Cross-over_Cable |
| 13:20.37 | Dovid | i assume thats a T1/E1 cross? |
| 13:23.11 | *** join/#asterisk hypknight (~hah@unaffiliated/hypknight) |
| 13:25.20 | hypknight | question for you guys: I'm running Asterisk 1.8 and watching events that are firing out of AMI... I'm looking for an event to signify the login/logout of an agent from a queue using Queue Toggle... I'm seeing the extension go off-hook, ring, and hangup when through 'Newstate's when the agent dials *45, but I don't see any events about the queues... AMI is setup with this string: all,system, |
| 13:25.20 | hypknight | call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate -- Any ideas? |
| 13:25.21 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 13:25.26 | danfromuk | Both the sip trunk and the sip peer are both set to allow=gsm, but ulaw sounds are still being used. Any ideas? |
| 13:26.36 | MrTelephone | I tried to collect a bunch of logs for the t38 problems I'm having. There is too much information on the production machine to sift through :(. The modem re-invites with the t38 sdp information. Asterisk sends a sip 100 TRYING and adds a t38_abort to the scheduler at a time of 5000ms. In the logs the 488 Not Acceptable sip message is sent from asterisk after exactly 5 seconds. Asterisk does |
| 13:26.36 | MrTelephone | not send a sip t38 sdp response after the 100 trying message. I looked at the source code and it looks like because there is no AUDIO parameter in the modem reinvite message asterisk does not send a t38_sdp message. Is there supposed to be an rtpmap in the t38 message from the endpoint? |
| 13:27.38 | MrTelephone | I get an error message saying "no audio, accepting t38 request anyways" but the source code does not tell asterisk to send a t38 sdp message. |
| 13:29.02 | MrTelephone | I did a tcpdump and I seen asterisk sending packets to the proper t38 port towards the endpoint but no traffic from the endpoint is shown towards asterisk. I'm assuming this is because of the lack of t38 parameters from asterisk. |
| 13:31.36 | WIMPy | Dovid: Yes, that's the one. |
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| 13:46.35 | Dovid | wimpy: That was the issue. the cross fixed it |
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| 13:49.19 | elliot98 | gives a wave to all |
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| 13:49.41 | danfromuk | ive deleted all the asterisk sounds. i'm trying make install to bring them back but it doesnt seem to help. any ideas? |
| 13:50.12 | elliot98 | danfromuk: are file permissions properly set? |
| 13:50.18 | jacc0 | make menuselect |
| 13:50.23 | jacc0 | and then select them |
| 13:50.24 | danfromuk | yes, and i'm not getting any errors. |
| 13:50.34 | danfromuk | in menuselect, the ones i want are selected. |
| 13:50.58 | jacc0 | then do make clean |
| 13:51.04 | jacc0 | followed by make install |
| 13:51.07 | elliot98 | what kind of T38 support does Asterisk now have...is it native or would a third-party add-on still be necessary? |
| 13:51.42 | jacc0 | (you need internet connection) |
| 13:51.51 | danfromuk | i do |
| 13:52.01 | danfromuk | its connected, i'm working remotely |
| 13:52.24 | jacc0 | where do you look for the soundfiles? |
| 13:52.57 | danfromuk | var/lib/asterisk/sounds (with a / at the start. irc wont let me put the first /) |
| 13:53.25 | jacc0 | I get it |
| 13:53.39 | danfromuk | nope. make clean and make install didnt download the sound files. |
| 13:53.43 | jacc0 | do you have write promission in that folder? |
| 13:53.54 | danfromuk | yes, i'm installing as root. |
| 13:54.03 | danfromuk | there must be a flag somewhere saying that they have already been downloaded. |
| 13:54.14 | jacc0 | I believe it won't re-download if they are already there |
| 13:54.15 | danfromuk | i've deleted the sounds folder to see if would help, but it didnt |
| 13:57.31 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 13:59.23 | *** join/#asterisk din3sh (din3sh@41.136.246.205) |
| 13:59.47 | wdoekes2 | danfromuk: ~/.asterisk_sounds_cache ? |
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| 14:00.13 | danfromuk | i ran ./configure first which seems to have got it to redownload the sounds |
| 14:01.26 | danfromuk | argh. where would a .slin sound file be stored? i've tried to force asterisk to only use gsm by deleting all other sound files, but its still finding a slin sound file from somewhere. |
| 14:01.56 | Jasnejac | danfromuk: /var/lib/asterisk/sounds/en |
| 14:02.21 | danfromuk | that folder only contains .wav and .gsm files |
| 14:02.34 | Jasnejac | has to be there somewhere |
| 14:03.03 | danfromuk | agreed, but where? |
| 14:03.14 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 14:03.20 | danfromuk | how does asterisk decide which folder to use? |
| 14:03.20 | Jasnejac | what file is it playing? what announcement? |
| 14:04.11 | danfromuk | <PROTECTED> |
| 14:06.14 | *** join/#asterisk RubyRails (~justin@209.33.214.243) |
| 14:06.20 | Jasnejac | that should be in the root sounds directory |
| 14:06.21 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
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| 14:08.42 | danfromuk | dir *.slin shows up nothing |
| 14:08.57 | Jasnejac | try *.sln |
| 14:09.29 | danfromuk | <PROTECTED> |
| 14:10.13 | Jasnejac | what formats are there in there? |
| 14:11.27 | danfromuk | ive been having a problem with voicemail, so just gsm and wav to try to sort it out |
| 14:15.30 | *** join/#asterisk mpe_ (~mpe@109.70.54.56) |
| 14:16.35 | *** join/#asterisk mpe (~mpe@109.70.54.56) |
| 14:17.45 | Jasnejac | has anyone managed to get the interface to samngoma cards working from a virtualised environment? |
| 14:19.02 | MrTelephone | Can someone take a look at my T38 failure at http://pastebin.com/VxPYsi6p ? |
| 14:21.23 | danfromuk | is disallow=all and allow=gsm why do I get this? |
| 14:21.24 | danfromuk | <PROTECTED> |
| 14:21.24 | danfromuk | <PROTECTED> |
| 14:21.35 | danfromuk | <PROTECTED> |
| 14:21.35 | danfromuk | <PROTECTED> |
| 14:22.22 | file | MrTelephone, you should post the complete console log... not a part |
| 14:23.10 | MrTelephone | from beginning of the call to the end? |
| 14:23.16 | file | yes |
| 14:23.23 | MrTelephone | ok |
| 14:23.42 | wdoekes2 | danfromuk: wav is SignedLiNear |
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| 14:24.56 | danfromuk | ah, ok. |
| 14:25.07 | danfromuk | so now i'm left with my voicemail issue. |
| 14:25.16 | danfromuk | ive just upgraded from 1.4 to 1.8 and i have an issue with the voicemail. If the call goes straight to the voicemail app it works fine. But if the call goes to a sip peer, times out and goes to the voicemail app, I hear alison as normal, then the beep, but the beep seems to continue and it doesnt record properly. |
| 14:28.30 | coppice | wdoekes2: wav is just a contain, many things (including signed linear) and be in it |
| 14:28.45 | *** join/#asterisk mealstrom (~mealstrom@46.63.71.254) |
| 14:29.35 | *** part/#asterisk mirelab (~mirko@212.200.146.253) |
| 14:33.55 | wdoekes2 | coppice: true. but in 99% of the cases it is slin I think, not to mention in this case |
| 14:35.35 | elliot98 | how reliable is t38 in *1.8? |
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| 14:38.35 | danfromuk | in 1.8, alison seems to have stopped saying 'please leave you message after the tone'. is there any way to turn that back on? |
| 14:41.52 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 14:42.57 | [TK]D-Fender | danfromuk, It gives instructions if you tell it to give instructions |
| 14:43.55 | danfromuk | sorry, didnt spot that i was using 's'. |
| 14:44.09 | danfromuk | thought it was related to the problem above. |
| 14:46.46 | danfromuk | ok, i resolved the above beep issue by deleting all sounds files and only installing gsm files. |
| 14:47.02 | danfromuk | looks like there is a codec transformation problem somewhere. |
| 14:47.44 | *** join/#asterisk sustav (~alfa@nat/digium/x-yljqotbwfisdsfmu) |
| 14:52.22 | danfromuk | wdoekes2: thanks for pointing out that slin meant wav. I would have been looking for a .slin file for hours! |
| 14:52.32 | MrTelephone | I have a feeling asterisk will not send t38 sdp without faxdetect=on |
| 14:53.39 | MrTelephone | But I have it turned on |
| 14:55.16 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 14:57.09 | eduzimrs | Hi, anyone knows about a problem using TLS and SRTP in * 1.8.11 ? When i hangup the call it doesnt hangup at asterisk, the channel continues UP until RTP inactivity time is reached |
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| 15:01.44 | MrTelephone | Even after asterisk tries to negotiate and fails the t38 fax counters don't increment. Negotiation Failed : 0 |
| 15:02.03 | MrTelephone | Does fax for asterisk even work for version 10+? |
| 15:02.09 | leifmadsen | yes |
| 15:02.33 | MrTelephone | Is setting the VAR in dialplan mandatory for it to work? |
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| 15:19.40 | file | MrTelephone, if you are trying to do gatewaying (one side SIP using T38, one side DAHDI) you need to use spandsp as only it supports it |
| 15:20.19 | file | I was going to use the full log to confirm exactly what you are doing, but I'll just throw out the above |
| 15:20.57 | MrTelephone | I'm trying to get a log. It's almost a megabyte. Will that fit in pastebin? Is there somewhere I can attach a file? |
| 15:21.24 | file | let's just go for the question approach then |
| 15:21.38 | MrTelephone | DAHDI/33-1 is trying to dial SIP/2298035. 2298035 t38 goes to state 2 and dahdi goes off hook as usual. |
| 15:21.47 | file | so you are doing T.38 fax gateway |
| 15:22.12 | MrTelephone | I'm almost positive it worked before but I cannot say what version it quit working on. I'm trying to do 1 leg t38 to the pstn gateway asterisk machine. |
| 15:22.23 | file | have you installed spandsp and recompiled Asterisk so it has the spandsp fax module? |
| 15:22.39 | MrTelephone | I will confirm. I purchased fax for asterisk and assumed that was all I needed. |
| 15:22.49 | file | Fax for Asterisk does not support gatewaying |
| 15:22.58 | coppice | fax for asterisk won't work. you need spandsp |
| 15:24.00 | MrTelephone | is FaxReceive and FaxSend apps part of spandsp? |
| 15:24.56 | coppice | spandsp does everything. fax for asterisk only does endpoint faxing |
| 15:25.14 | file | do you mean SendFAX and ReceiveFAX? if so, no - those are applications in Asterisk which utilize an interface to different modules (fax for asterisk OR spandsp) |
| 15:25.50 | file | the T38 gateway support does the same thing, but fax for asterisk does not implement the interface required for gatewaying so you have to use spandsp |
| 15:26.07 | MrTelephone | What I don't need to do is terminate faxes at the gateway. I'll install spandsp and try again. |
| 15:29.19 | *** join/#asterisk scgm11 (~Sebastian@r186-50-87-24.dialup.adsl.anteldata.net.uy) |
| 15:29.45 | scgm11 | hi, is it possible to transfer a ringing channel?? |
| 15:30.02 | scgm11 | I have the phones with answer at the beginning of the calls so they can pass dtmf |
| 15:30.11 | scgm11 | but when I use those dtmf |
| 15:30.25 | scgm11 | they are not being selected as feature |
| 15:30.27 | scgm11 | for the transfer |
| 15:36.57 | scgm11 | anyone? |
| 15:37.36 | WIMPy | That really depends on the phone. |
| 15:38.24 | scgm11 | why on the phone? if asterisk took the dtmf before the bridge this should work?? |
| 15:39.08 | WIMPy | You can;t use them befor you answer the call for obvious reasons. |
| 15:40.02 | scgm11 | I really don't see the obvious here, any conventional pbx allow transfer ringing calls... |
| 15:40.32 | WIMPy | Which is done by the phones. |
| 15:40.46 | WIMPy | And Asterisk is not a conventional PBX. |
| 15:40.59 | MrTelephone | setvar=FAXOPT(gateway)=yes <-- is this important for sip t38 to terminate with asterisk for transmission via dahdi span? |
| 15:41.00 | scgm11 | I tried with d option on dial so I make a redirect of the ringing channel |
| 15:41.15 | ZenMaster | Morning gentlemen. |
| 15:41.17 | WIMPy | What kind of phone are you using? |
| 15:41.34 | scgm11 | Yealink, Linksys and Snom |
| 15:41.37 | WIMPy | Evening ZenMaster |
| 15:42.17 | WIMPy | scgm11: At least the snom can do it when you have the destination on a button. Haven't tried without. |
| 15:42.29 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 15:42.32 | WIMPy | But it has to be enabled. Don;t remember what the default is. |
| 15:43.17 | puzzled | hi |
| 15:43.22 | WIMPy | Works perfectely without buttons. |
| 15:44.08 | scgm11 | WIMPy: so something has to be enable on snom and use the transfer button right?? not the asterisk feature? |
| 15:44.13 | WIMPy | For Linksys I'd almos bet it doesn't work. And no idea about Yealik. They didn't cross my way. |
| 15:45.02 | WIMPy | scgm11: To do it via a dest button, yes. To do it via the transfer button, no. |
| 15:45.33 | scgm11 | ok so only works via dest button? |
| 15:45.53 | file | MrTelephone, if you want it to work yes |
| 15:47.03 | WIMPy | scgm11: No. |
| 15:47.17 | MrTelephone | Should I see some activity then when I do 'fax show stats' ? |
| 15:47.29 | MrTelephone | I already did the recompile and ready to give it a shot |
| 15:47.39 | WIMPy | scgm11: It should always work via transfer button. And optionally via dest buttons. |
| 15:47.46 | MrTelephone | I have to add setvar column to my realtime database |
| 15:48.27 | WIMPy | And at least the old Linksys can't do it. If you click on forward while it's ringing, it will set up forwarding for following calls, keeping the current one ringing. |
| 15:49.16 | scgm11 | ok is not a transfer is a forward |
| 16:01.41 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
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| 16:17.12 | *** join/#asterisk AceError (~quassel@193.27.192.2) |
| 16:21.52 | AceError | Hi everone! I need some help. Is there anyone feeling helpful today? I have an extesion that when in rings on the phone, its not possible to answere the call. I press the button to answer but it continues to ring. It just that extension, all the others work. |
| 16:22.07 | *** part/#asterisk scgm11 (~Sebastian@r186-50-87-24.dialup.adsl.anteldata.net.uy) |
| 16:22.41 | eduzimrs | Hi, anyone knows about a problem using TLS and SRTP in * 1.8.11 ? When i hangup the call it doesnt hangup at asterisk, the channel continues UP until RTP inactivity time is reached |
| 16:26.29 | MrTelephone | tcpdump the connection to see what is going on |
| 16:26.43 | MrTelephone | is your endpoint sending data, maybe asterisk doesn't know what to do with it |
| 16:29.56 | MrTelephone | http://www.remiphilippe.fr/2011/01/16/asterisk-srtp-with-1-8/ |
| 16:30.57 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 16:30.57 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:36.01 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 16:38.17 | fubada | hi |
| 16:38.55 | fubada | im having some rtp issues with my provider nexvortex, its set to communicate with my pbx at udp 10000:20000, but i get incoming rtp outside of that range |
| 16:39.08 | fubada | is it generally safe to open all udp >1024 to the asterisk box |
| 16:40.19 | [TK]D-Fender | fubada, Show us your call and RTP debug |
| 16:40.24 | *** join/#asterisk PBXman (c335d9a5@gateway/web/freenode/ip.195.53.217.165) |
| 16:40.31 | PBXman | hello Asterisk people |
| 16:40.37 | fubada | [TK]D-Fender: i would but i can just simply show you my tcpdump |
| 16:40.41 | fubada | 12:38:05.983743 IP sip-eng1.appiaservices.com.64730 > ool-60395c1c.static.optonline.net.26874: UDP, length 172 |
| 16:40.55 | fubada | that is well outside of the specified 10000-20000 range on my provider side |
| 16:41.29 | fubada | [TK]D-Fender: whenever it falls within range, everything works |
| 16:41.33 | fubada | whenever it doeesnt, one way audio |
| 16:41.36 | rspp2k | hello PBXman |
| 16:41.42 | fubada | dropping iptables fixes the call |
| 16:41.57 | PBXman | has anybody got a Digium Asterisk Certification? |
| 16:42.18 | [TK]D-Fender | fubada, Show us the configs and SIP debug to match |
| 16:42.36 | fubada | ok [TK]D-Fender you know im bad at this, can you clarify what you need me to show |
| 16:42.37 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:42.41 | [TK]D-Fender | PBXman, Next non-meta question please... |
| 16:42.44 | fubada | also, it happens every few calls |
| 16:42.48 | fubada | so it'll be hard to capture |
| 16:42.57 | [TK]D-Fender | fubada, SIP debug for the call like I jsut said, and your rtp.conf |
| 16:43.08 | fubada | ok, i dont think i have rtp.conf at all |
| 16:43.30 | fubada | yep, no rtp conf, and Im working on the sip debug |
| 16:43.31 | [TK]D-Fender | that = fail |
| 16:44.29 | fubada | i copied the rtp.conf from my orig config |
| 16:44.57 | PBXman | <PROTECTED> |
| 16:45.26 | [TK]D-Fender | PBXman, Do you see places you care about asking for it? |
| 16:45.58 | PBXman | Not really [TK]D-Fender , it just look good on your CV don't you think |
| 16:46.13 | [TK]D-Fender | Anything more = looks good on CV |
| 16:47.10 | rspp2k | CV's that catch my eye show projects the person has completed or been involved with. What certs and where you went to schools means little to me. |
| 16:48.09 | PBXman | agreed rspp2k but not everybody who may interview you agrees with you |
| 16:48.13 | rspp2k | instead of spending time on a certification I would create my own project and complete it |
| 16:48.51 | rspp2k | true. A wise person doesn't dismiss experience. |
| 16:49.01 | rspp2k | I prefer to be hired by wise people. |
| 16:49.01 | _Corey_ | Someone having a dCAP would influence a hiring decision with me... an MS cert, not so much |
| 16:50.07 | _Corey_ | Digium doesn't hand them out to any idiot who can regurgitate crap from a book... they have to actually demonstrate proficiency |
| 16:50.31 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:50.35 | WIMPy | ... on historic technology. |
| 16:52.03 | Jasnejac | what's the betting there will be a new cert for the digium phones? |
| 16:52.17 | rspp2k | The only reason I have seen to get certifications is for Cisco so you can meet the requirements for their roles that are required for channel partnership. |
| 16:52.30 | [TK]D-Fender | Jasnejac, slim/none |
| 16:53.51 | fubada | [TK]D-Fender: http://pastebin.ca/2152359 http://pastebin.ca/2152358 |
| 16:53.59 | fubada | rtp.conf and sip debug |
| 16:54.52 | fubada | in this case the provider hit my pbx on port 9144 for rtp (outside of the 10000-20000 range specified in rtp.conf and provider side) |
| 16:59.17 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 17:01.20 | [TK]D-Fender | fubada, m=audio 9144 RTP/AVP 0 101 |
| 17:01.41 | [TK]D-Fender | fubada, "ls -la /etc/asterisk |
| 17:02.07 | [TK]D-Fender | fubada, and since you said the file was missing entirely, did you restart * completely upon restoring a config? |
| 17:02.34 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 17:02.48 | fubada | yep, i added the file and restarted asterisk using the init script |
| 17:03.21 | [TK]D-Fender | fubada, "ls -la /etc/asterisk |
| 17:03.31 | fubada | yes, i see that my umask |
| 17:03.33 | fubada | 0002 |
| 17:03.48 | fubada | thanks, asteriks didnt see the file |
| 17:06.44 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:11.05 | *** join/#asterisk navaismo (~navaismo@201.123.94.189) |
| 17:14.21 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:16.07 | *** join/#asterisk dddh (~dddh@pdpc/supporter/active/dddh) |
| 17:22.56 | MrTelephone | how do you reset interface counters for sangoma |
| 17:25.28 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 17:28.02 | leifmadsen | _Corey_: +1 |
| 17:28.28 | leifmadsen | _Corey_: test is basic installation and common knowledge, the hard part being that you have limited time to perform the practical portion of the test :) |
| 17:29.04 | MrTelephone | For what interview? |
| 17:30.07 | MrTelephone | Do you now see any sip applications for playbook because there is no sip development packages? |
| 17:30.32 | MrTelephone | It's got to be an audio issue because basic sip dialog shouldn't be too hard to make |
| 17:33.16 | fubada | im missing the meetme application, can someone tell me which module to load |
| 17:34.00 | [TK]D-Fender | fubada, DAHDI <- |
| 17:34.17 | [TK]D-Fender | fubada, needs to be properly set up prior to * |
| 17:34.30 | fubada | thanks i dont use dahdi so that is why |
| 17:36.24 | MrTelephone | Someone needs to make a tablet that supports windows 8 |
| 17:36.27 | MrTelephone | :( |
| 17:37.32 | [TK]D-Fender | The will... sometime after Windows 8 actually exists... |
| 17:38.46 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:40.20 | fubada | [TK]D-Fender: i installed dahdi modules, and now I see MeetMe with 'core show applications' |
| 17:40.36 | fubada | however, when i dial in to my test meetme conf, I get a dropped call |
| 17:40.56 | fubada | http://pastie.org/3951134 |
| 17:40.57 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 17:41.11 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 17:41.20 | [TK]D-Fender | Make sure dahidh is initialised before you start * |
| 17:41.38 | fubada | how do i do that :) |
| 17:42.13 | MrTelephone | dahdi_cfg -vv |
| 17:42.28 | fubada | well i dont have any dahdi devices |
| 17:42.39 | MrTelephone | there should be one with the word dummy in it |
| 17:43.00 | fubada | http://pastie.org/3951139 |
| 17:43.03 | MrTelephone | or did they ditch that |
| 17:43.13 | *** join/#asterisk atmark (~mark@fedora/atmark) |
| 17:43.49 | MrTelephone | modprobe dahdi_dummy and see what happens. When you do lsmod do you have any dahdi drivers loaded at all? |
| 17:44.07 | atmark | Newbie questions, is asterisk an SIP server? |
| 17:44.14 | [TK]D-Fender | atmark, Not really. |
| 17:44.17 | [TK]D-Fender | ~b2bua |
| 17:44.17 | infobot | well, b2bua is a Back 2 Back User Agent. Additional information is available on wikipedia: http://en.wikipedia.org/wiki/Back-to-back_user_agent |
| 17:44.19 | [TK]D-Fender | ^^^ |
| 17:44.24 | fubada | no such module, and no loaded dahdi modules of any kind |
| 17:44.31 | MrTelephone | did you make install? |
| 17:44.38 | fubada | used epel rpm's |
| 17:44.39 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 17:44.50 | MrTelephone | modprobe dahdi? |
| 17:44.57 | navaismo | vomit |
| 17:45.33 | atmark | How does asterisk differ from a SIP Server? |
| 17:45.58 | MrTelephone | atmark, it is what you want to call it. asterisk supports sip protocol |
| 17:46.07 | MrTelephone | that's all you need to know |
| 17:46.16 | MrTelephone | sip phone + asterisk = working |
| 17:47.17 | [TK]D-Fender | atmark, "sip server" is a vague term. Could mean any of several functions, include SIP Proxy (* is not), RTP proxy (* is not), interlinked registration server (* is not) |
| 17:47.19 | atmark | thanks I understand it a bit. |
| 17:47.25 | [TK]D-Fender | atmark, What do you actually require? |
| 17:48.31 | atmark | No I'm just curious on how asterisk works and what components does it needs |
| 17:49.00 | [TK]D-Fender | atmark, * cal talk to all sorts of equipement and protocols. |
| 17:49.03 | [TK]D-Fender | atm |
| 17:49.05 | [TK]D-Fender | ... |
| 17:49.06 | [TK]D-Fender | ~book |
| 17:49.06 | 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://www.asteriskdocs.org/ or see ~buybook |
| 17:49.08 | [TK]D-Fender | ^^^^ |
| 17:49.12 | [TK]D-Fender | aLL DESCRIBED IN THERE |
| 17:49.20 | [TK]D-Fender | dang caps... |
| 17:49.21 | fubada | MrTelephone: there doesnt appear to be any dahdi dummy drivers |
| 17:49.35 | atmark | wow, you type fast |
| 17:49.36 | MrTelephone | interesting |
| 17:49.36 | atmark | thanks |
| 17:49.43 | fubada | MrTelephone: is dahdi necessary for MeetMe? |
| 17:49.49 | MrTelephone | I though it was |
| 17:49.54 | fubada | tk said it was |
| 17:49.55 | MrTelephone | Fender confirmed it |
| 17:50.08 | MrTelephone | yeah he knows what he is talking about 99% of the time |
| 17:50.36 | MrTelephone | dahdi comes with a bunch of hardware drivers for digium analog/digital cards. If you don't have a card you can still load the modules |
| 17:50.46 | MrTelephone | try insmod dahdi_dummy |
| 17:50.47 | MrTelephone | or something |
| 17:51.24 | MrTelephone | If you don't know how to load a kernel driver you should do some playing around on your shell and figure it out |
| 17:51.42 | MrTelephone | dahdi - wanpipe,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp |
| 17:51.46 | fubada | theres just no such module .ko fil |
| 17:51.48 | fubada | e |
| 17:52.07 | fubada | i search with find / -name *dahdi*.ko, nothing |
| 17:52.09 | MrTelephone | then you probably installed the tools, not the drive package |
| 17:52.41 | leifmadsen | MeetMe() requires DAHDI, not just for timing, but for the actual audio mixing as well (so other timing sources are not sufficient for MeetMe() |
| 17:53.00 | fubada | thanks guys |
| 17:53.10 | leifmadsen | MrTelephone: you don't need to explicitly load dahdi_dummy |
| 17:53.12 | fubada | MrTelephone: http://pastie.org/private/5otttbguefj4edngbwlua all of those are installed in x86_64 |
| 17:53.12 | MrTelephone | check your rpm again and make sure you got the drivers in there |
| 17:53.14 | leifmadsen | that is built into dahdi core now |
| 17:53.25 | leifmadsen | just loading DAHDI is enough -- no drivers necessary |
| 17:53.31 | MrTelephone | nice |
| 17:53.37 | leifmadsen | has been like that for a few months |
| 17:53.39 | fubada | leifmadsen: i dont need the kernel drivers? |
| 17:53.52 | MrTelephone | you need to load the main dahdi module he is saying |
| 17:53.54 | leifmadsen | if you just need it for timing and meetme() mixing, you just need dahdi loaded |
| 17:54.05 | fubada | ok, i think it should be auto loaded |
| 17:54.23 | fubada | in my modules.conf i have "noload => app_dahdiras.so |
| 17:54.29 | MrTelephone | ./lib/modules/2.6.32-5-686/dahdi/dahdi.ko |
| 17:54.31 | fubada | is that dahdi disabled? |
| 17:54.36 | MrTelephone | no |
| 17:54.39 | MrTelephone | kernel module |
| 17:54.45 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 17:54.49 | fubada | ah ok |
| 17:54.55 | leifmadsen | res_timing_dahdi.so |
| 17:55.03 | MrTelephone | you should have a dahdi fodler in your /lib/modules/<kernel version>/ folder |
| 17:55.08 | leifmadsen | app_dahdiras.so is an application (hence the app_ prefix) |
| 17:55.17 | leifmadsen | are you compiling or RPMing? |
| 17:55.22 | MrTelephone | RPMing he said |
| 17:55.26 | fubada | http://pastie.org/private/xs4tm2eyizxoyij1zhosuq |
| 17:55.29 | leifmadsen | steps away |
| 17:55.31 | fubada | no such thing mang, rom |
| 17:55.35 | fubada | rpm |
| 17:55.51 | *** join/#asterisk mroe (~mroe@unaffiliated/roe) |
| 17:55.55 | fubada | MrTelephone: what repo did your dahdi driver come from? |
| 17:55.57 | MrTelephone | what was the rpm called |
| 17:56.09 | MrTelephone | I download them from asterisk site because distro seem to be slow for updates |
| 17:56.13 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-mlnvgnhbrhefrprh) |
| 17:56.14 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 17:56.14 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 17:56.31 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 17:56.31 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 17:56.38 | MrTelephone | better off to download the source and compile and you'll be on the same page as most people in here I think |
| 17:56.40 | *** join/#asterisk serafie (~erin@nat/digium/x-nsgseiqastnrpjyl) |
| 17:57.15 | MrTelephone | when you ask for support you'll notice that if you have an old version noone wants to help until you upgrade |
| 17:57.27 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-mhztfrknfiiptknw) |
| 17:57.27 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 18:04.06 | *** join/#asterisk dandate2 (~dan@180.190.210.10) |
| 18:04.15 | dandate2 | what command do I use to see how many g729 licenses are installed? |
| 18:05.14 | [TK]D-Fender | g729 show |
| 18:05.34 | fubada | awsome dahdi fixed |
| 18:05.46 | MrTelephone | how did you do it |
| 18:05.56 | fubada | wonder if the drivers built from source will survive a kernel upgrade |
| 18:06.03 | fubada | MrTelephone: built from source 2.6.1 |
| 18:06.13 | MrTelephone | no they need to be recompiled |
| 18:08.27 | MrTelephone | I haven't had an updated kernel for some time with debian stable |
| 18:16.27 | carrar | 3.2.18 with btrfs! |
| 18:16.36 | carrar | heh |
| 18:17.29 | dandate2 | does g729 sound better than gsm if no packet loss? |
| 18:17.42 | carrar | yes |
| 18:29.29 | *** part/#asterisk ndespres (~ndespres@208.103.80.226) |
| 18:29.52 | *** join/#asterisk din3sh (din3sh@41.136.244.231) |
| 18:31.58 | MrTelephone | "did you pull the seal strip out of the toner cartridge before you started printing?" |
| 18:32.00 | MrTelephone | my gawd |
| 18:32.32 | MrTelephone | worst part is you get the attitude, "well it worked good before the IT department moved my computer around" |
| 18:32.41 | MrTelephone | no, it didn't work right since you put the new cartridge in |
| 18:32.56 | MrTelephone | the human race is doomed |
| 18:34.10 | jpsharp | And once you work on someone's computer, *every* problem they have from there on out is your fault. |
| 18:34.30 | din3sh | MrTelephone: call your System Administrator :p |
| 18:34.44 | fubada | this is why hosting a telephony solution is always a bad idea |
| 18:34.46 | fubada | also email |
| 18:34.56 | fubada | youll be blamed for every little hiccup |
| 18:35.00 | MrTelephone | I'm calling the CEO |
| 18:35.58 | din3sh | CEO cum system admin ;) |
| 18:36.34 | *** join/#asterisk miztic (~miztic@75-149-203-105-Illinois.hfc.comcastbusiness.net) |
| 18:37.00 | MrTelephone | barrack.obama@whitehouse.us.gov |
| 18:37.07 | MrTelephone | email him and tell him our concerns |
| 18:37.44 | MrTelephone | If you phone system does not work exactly like traditional telephone then you may run into a few snags |
| 18:40.20 | d_preston215 | Any sure fire way to kill a stuck channel? |
| 18:40.53 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
| 18:41.58 | paulc | "core restart now"? ;-) |
| 18:42.34 | *** join/#asterisk jm_s2s (~jm_work_s@static-72-64-129-133.tampfl.fios.verizon.net) |
| 18:43.26 | MrTelephone | I thought there was a channel destroy command? |
| 18:43.49 | *** join/#asterisk kenshin (~kenshin@dsl-173-248-237-61.acanac.net) |
| 18:44.21 | jpsharp | For the dahdi driver, yes. |
| 18:44.28 | akrohn | greetings. Does anyone know how to update my CDR() after a Dial(SIP/1&SIP/2&SIP/3,10,M(cdr_update)) command to let me know which phone answered the call? Set(CDR(foo)=bar) is executing but won't stick. This seems to be a known problem. Does anyone know a workaround? |
| 18:45.07 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:46.05 | d_preston215 | Internal ChanSpy SIP channel thats zombified. |
| 18:46.33 | [TK]D-Fender | AMI redirect it off a cliff |
| 18:47.04 | MrTelephone | tnt_zombie.exe |
| 18:47.21 | akrohn | cool thanks for the info! I'll see what i can come up with |
| 18:48.14 | jpsharp | There's also the "hangup request" command, but that's not always guaranteed to nuke the channel |
| 18:55.32 | dandate2 | i just purchased more g729 licenses and already had some installed. what is the command to add more licenses? |
| 18:58.09 | newtonr | dandate2: http://downloads.digium.com/pub/telephony/codec_g729/README should be able to just run the register utility with your new license , but may have to reload asterisk |
| 18:59.15 | newtonr | dandate2: can probably just reload the codec_g729a.so module after the new registration, IIRC |
| 18:59.26 | dandate2 | k |
| 19:00.40 | *** join/#asterisk Jinxed- (536f6c82@gateway/web/freenode/ip.83.111.108.130) |
| 19:02.05 | Jinxed- | Odd question. I'm trying to connect my android phone to my network to register with asterisk. I'm trying to connect the phone via usb (IP over USB) with android, has anyone ever tried that here. I know it's not technically asterisk related but figured someone might point me in the right direction. |
| 19:02.35 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 19:02.35 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 19:10.41 | jpsharp | IP over USB. That sounds just...well...wrong. |
| 19:12.13 | *** join/#asterisk sarsaeol (~sarsaeol@66-113-78-49.rev.ibsinc.com) |
| 19:13.41 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
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| 19:17.40 | MrTelephone | IP over USB is popular in the isp industry |
| 19:17.58 | *** join/#asterisk scgm11 (~Sebastian@r186-50-87-24.dialup.adsl.anteldata.net.uy) |
| 19:18.16 | scgm11 | is it possible to use ChannelRedirect on a ringing channel?? |
| 19:18.58 | jpsharp | MrTelephone: Really? HOw is it better than IP over Ethernet? |
| 19:19.57 | jpsharp | Or are you being sarcastic? |
| 19:20.04 | din3sh | scgm11: transfer a call without actually answering? |
| 19:20.12 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:20.14 | scgm11 | basically |
| 19:20.30 | scgm11 | I make an outgoing call, then while is ringing I want to transfer it |
| 19:20.36 | scgm11 | it doesn't work with features |
| 19:20.42 | *** join/#asterisk greenwolf (~Penguin@173.254.254.104) |
| 19:20.53 | scgm11 | but I think could be done with redirectchannel |
| 19:21.11 | scgm11 | using d option on dial to some part of the dial plan using channelredirect |
| 19:21.58 | *** join/#asterisk greenwolf (~Penguin@173.254.254.104) |
| 19:22.08 | din3sh | works with cisco 79xx series |
| 19:22.15 | din3sh | at least i've tested |
| 19:22.27 | scgm11 | what exactly?? |
| 19:22.52 | din3sh | <scgm11> I make an outgoing call, then while is ringing I want to transfer it |
| 19:23.13 | scgm11 | using features??? |
| 19:23.29 | scgm11 | or using the phone transfer?? |
| 19:27.55 | din3sh | phone |
| 19:28.11 | din3sh | sip refer |
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| 19:51.41 | talntid | http://pastebin.com/wKpgTDBS |
| 19:52.16 | talntid | I assume the answer is no, but... anyone know of a way to rename this monitor file with the "HUMAN" or "MACHINE", after the AMD()? |
| 19:52.23 | talntid | but I need it to start recording prior to the AMD |
| 19:53.07 | talntid | ah, ChangeMonitor! |
| 19:56.25 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 19:56.27 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 19:57.12 | talntid | [May 22 12:55:57] WARNING[4412]: file.c:505 ast_filehelper: rename(/mnt/storagedrive/Erics Awesome Place--20120522-125546-out.wav,/var/spool/asterisk/monitor/wav,/mnt/storagedrive/Erics Awesome Place-HUMAN-20120522-125549,m-out.wav) failed: Invalid cross-device link |
| 20:00.32 | *** part/#asterisk scgm11 (~Sebastian@r186-50-87-24.dialup.adsl.anteldata.net.uy) |
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| 20:11.52 | pabelanger | leifmadsen: where is the information about your hotdesking stuff for asterisk? |
| 20:12.07 | leifmadsen | pabelanger: Relational Database Integration chapter |
| 20:12.17 | pabelanger | ~book |
| 20:12.17 | 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://www.asteriskdocs.org/ or see ~buybook |
| 20:12.25 | pabelanger | ~hotdesking |
| 20:12.48 | leifmadsen | http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/getting_funky.html |
| 20:13.37 | pabelanger | ODBC eh? |
| 20:13.40 | pabelanger | hmm |
| 20:13.48 | leifmadsen | aye |
| 20:14.16 | pabelanger | can we do astdb via ODBC yet? |
| 20:14.33 | leifmadsen | nope |
| 20:15.03 | leifmadsen | you can do it with AstDB, just different logic |
| 20:15.07 | leifmadsen | (I've done it before) |
| 20:16.38 | pabelanger | Yar, I might just start with astdb |
| 20:22.05 | zamba | is there a place i can download different voice prompts from? i'm looking for norwegian voice prompts in particular |
| 20:25.31 | zamba | nevermind, found something |
| 20:25.58 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 20:25.59 | [sr] | howdy |
| 20:26.17 | *** join/#asterisk Hive (~Hive@173-165-205-1-jacksonville.hfc.comcastbusiness.net) |
| 20:26.23 | zamba | howdy! |
| 20:26.59 | Hive | Does Asterisk 10 support 'MeetMeCount()'? The wiki has it listed ( https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Application_MeetMeCount ) but i'm not able to get it working |
| 20:28.48 | leifmadsen | define "not able to get it working" |
| 20:29.07 | Hive | [May 22 15:14:57] WARNING[5828]: pbx.c:4359 pbx_extension_helper: No application 'MeetMeCount' for extension (macro-joinConference, s, 2) |
| 20:29.24 | Hive | core show applications doesnt list MeetMeCount or MeetMe for that matter |
| 20:34.12 | Hive | so I'm not sure if meetme has been completely removed from Asterisk 10 in place of ConfBridge, or if I'm just 'doing it wrong' |
| 20:34.43 | thecardsmith | did you see it in "make menuselect" ? |
| 20:38.30 | Hive | ahh okay it was down under the deprecated section |
| 20:38.37 | Hive | guess it's time to migrate >_< |
| 20:45.52 | fubada | did snmp change much from 1.6 to 1.8? |
| 20:46.04 | fubada | ive got this zenoss asterisk plugin id like to try |
| 20:46.06 | fubada | but its for 1.6 |
| 20:46.42 | fubada | Hive: you also need dahdi |
| 20:46.46 | fubada | for meetme |
| 20:47.39 | fubada | was it silly to install 1.8 given theres '10' |
| 20:52.09 | Qwell | fubada: That depends. Do you need the features in Asterisk 10? |
| 20:52.21 | fubada | im browsing the changelog, trying to find it |
| 20:52.25 | fubada | the feature list |
| 20:53.09 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 20:54.34 | *** join/#asterisk tulcod (~auke@x032089.its-s.tudelft.nl) |
| 20:58.35 | fubada | can someone tell me if "res_phoneprov" is a useful tool for managing polycom configs |
| 20:58.49 | fubada | is that an asterisk-gui only feature? |
| 20:59.01 | leifmadsen | you can use it separate of the gui |
| 20:59.11 | tulcod | can i use MP3Player as Application in a callfile (/var/spool/asterisk/outgoing/)? |
| 20:59.13 | fubada | does it make sense to use this plugin? |
| 20:59.15 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:59.27 | tulcod | if so, why does my message end after 22 seconds? |
| 21:03.33 | *** join/#asterisk jsjc (~Adium@119.Red-2-136-85.dynamicIP.rima-tde.net) |
| 21:04.05 | Qwell | MP3Player is a terrible application, and nobody should use it. |
| 21:04.47 | GeoGeek | Having trouble getting a hint to work in 1.8.12. My BLF lights are working but I created a hint for a custom device using exten => 1001,hint,CUSTOM:ROBOT and Set(DEVICE_STATE(Custom:ROBOT)=INUSE) so I can have a button that lights to tell if we are in night service mode. Not working. Core show hints lists it but no subscriptions. |
| 21:05.24 | Qwell | well, figure out why nothing is subscribing to it |
| 21:06.36 | tulcod | Qwell: so Playback should work just fine? |
| 21:06.37 | fubada | leifmadsen: Qwell can you point me in the right direction? what do i need for phoneprov? dhcpd setting next-server to apache ? |
| 21:06.44 | fubada | er not leifmadsen, Qwell ^ |
| 21:07.00 | Qwell | tulcod: yes |
| 21:07.02 | Qwell | fubada: got me |
| 21:10.03 | GeoGeek | Qwell: Thanks for the suggestion. As luck would have it, I have already tried to figure that out but no luck so far. I was hoping for a ninja suggestion or a magic spell.... :-) |
| 21:10.10 | fubada | the http module needs to be used with phoneprov or can i use an external webserver such as apache? |
| 21:10.34 | Qwell | fubada: You have to use the http server in Asterisk |
| 21:10.50 | fubada | thank you |
| 21:11.26 | *** join/#asterisk emsLinux (~emsLinux@186.81.82.245) |
| 21:12.17 | emsLinux | Hello, do someone knows if there is any document where i can find the transcription of Asterisk's audio files so i can make my own using this guide? |
| 21:12.34 | Qwell | emsLinux: included with the sounds tarball |
| 21:15.09 | emsLinux | Qwell, are you talking about the asterisk tarball or there's another tarball with just the asterisk sounds? |
| 21:15.28 | Qwell | emsLinux: there are tarballs for just sounds |
| 21:15.56 | emsLinux | Qwell, great, are they in the Asterisk website, or where can i get them? |
| 21:17.50 | tulcod | Qwell: even with playback, the audio file is cut off after ~20 secs |
| 21:18.23 | tulcod | Qwell: i have "WaitTime: 15" in my callfile, but I thought that was the time till Answer |
| 21:23.30 | emsLinux | Qwell, found them, thank you for your quick answer |
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| 22:06.01 | fubada | astvarlibdir => /usr/share/asterisk |
| 22:06.06 | fubada | is this how this is supposed to be? |
| 22:06.15 | fubada | not /var/lib/asterisk? |
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