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| 03:15.51 | raden | how are all these people getting free long distwance with google vboice ? |
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| 03:18.15 | raden | why would they do that ? |
| 03:18.25 | raden | are they trying to kill the phone industry ? |
| 03:20.29 | raden | wow makes me hate googlew |
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| 03:37.37 | jamko | Google wants to kill a lot of things. Wal-Mart plague. |
| 03:53.41 | Cynagen | raden: what is the issue with free long distance? |
| 03:54.19 | Cynagen | you get it on cellular already with most carriers... most of the call is over some form of VoIP anyways |
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| 05:36.55 | desk003 | Hey, anyone happen to be around? I've a question about AsteriskNOW 2.0.2 w/ FreePBX? CDRs aren't showing, looks like cdr_adaptive_odbc isn't installed. What's the proper way to install it? |
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| 14:25.42 | Agro | Can anyone explain to me what CNG and CED tones are? |
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| 14:29.37 | ChannelZ | CNG is the tone FAX machines use to signal a fax.. 1100hz |
| 14:30.22 | ChannelZ | CED not sure |
| 14:31.38 | ChannelZ | here: http://telecom.tbi.net/fax-call.htm |
| 14:45.20 | Agro | ChannelZ, do you know what amp(amplitude) is? |
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| 16:12.24 | AndyHarris2 | Afternoon All ... Could someone let me know if there are any issues with alert-info and queues. I have it working for inbound routes and ring groups. However I can't get it working for static agents in a queue --- thanks. |
| 16:14.32 | *** join/#asterisk samfisher (~unaffilia@unaffiliated/samfisher) |
| 16:14.55 | samfisher | hi. Why everybody runs asterisk on CentOS? Could I use ubuntu or debian instead? |
| 16:17.50 | Agro | I run asterisk on freebsd |
| 16:19.52 | pabelanger | samfisher: you should be able to use any *NIX based OS |
| 16:20.01 | pabelanger | your support for asterisk will vary |
| 16:21.26 | samfisher | oh, so it's just support |
| 16:22.41 | samfisher | and if I make my own PBX with asterisk, I could create accounts and let my friend from another town to connect his IP phone to his router and we'll talk freely? |
| 16:22.48 | [TK]D-Fender | AndyHarris2: There isn't |
| 16:23.13 | samfisher | and if we add other friends, we can have a small telephony? |
| 16:23.35 | samfisher | telephony network... and make our own rules... IVR, voicemail etc |
| 16:23.51 | Cynagen | yes sam... that's kinda the idea |
| 16:24.25 | Cynagen | empower people to build their own telephony and start linking those together through services like enum if you have proper numbers |
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| 16:28.01 | AndyHarris2 | D-Fender --- did you mean there isn't support for alert-info in queues ? |
| 16:28.51 | AndyHarris2 | D-Fender --- or there aren't any issues and its expected to work ? |
| 16:29.51 | pabelanger | samfisher: yes |
| 16:32.12 | *** join/#asterisk catirerosy (~elanz@75-162-196-226.slkc.qwest.net) |
| 16:33.05 | Agro | Does anyone here know how to use the goertzel algorithm for tone detection? |
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| 16:42.11 | catirerosy | This may be a dumb question but I am new to asterisk.. Is asterisk suppose to create two wave files when a voicemail is placed under the /var/spool/asterisk/voicemail/default/extension#/INBOX? i.e msg000.WAV and msg000.wav |
| 16:46.25 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 16:51.29 | catirerosy | I am having an issue where voicemail messages are being converted from wav to mp3 files but the email voicemail attachment has 0 bytes... |
| 16:54.09 | kaldemar | catirerosy: those are different formats, your config enables both WAV and wav. |
| 16:54.56 | catirerosy | ok thanks kaldemar |
| 16:56.12 | catirerosy | do you know or can give me some guidance as to why the msg0000.mp3 files are being sent with 0 bytes? |
| 16:57.16 | carrar | Install Asterisk from Source |
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| 17:03.21 | catirerosy | where in the asterisk config can I tell it to only enable wav and not WAV |
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| 17:13.29 | catirerosy | <PROTECTED> |
| 17:14.37 | pabelanger | catirerosy: look at the format_ modules, depending on what you load, asterisk will use that format. EG: format_wav and format_wav_gsm |
| 17:14.50 | pabelanger | you all setup which formats you want voicemail to use in voicemail.conf |
| 17:15.07 | pabelanger | format=gsm |
| 17:15.23 | pabelanger | will save them as a gsm file |
| 17:18.55 | [TK]D-Fender | catirerosy: * can't transcode to MP3 something else is doing this and isn't part of * or the e-mailing process |
| 17:20.47 | catirerosy | thanks for your feed back guys, I am a linux admin but have 0 experience with asterisk and the phone admin that set it up left the company so I as assigned to take over :-) |
| 17:21.40 | catirerosy | I've been reading a lot and research google as well but can't figure out why the voicemail messages are being sent to users mail box with 0 bytes |
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| 17:22.39 | catirerosy | as soon as I call a number and leave a vm message I notice a long file format .wav file created under /var/spool/asterisk/monitor |
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| 17:23.24 | catirerosy | and under the /var/spool/asterisk/voicemail/default/ext#/INBOK a couple of files are created on WAV and one wav extension |
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| 17:24.01 | pabelanger | because you have format=wav,wav49 in voicemail.conf |
| 17:24.56 | catirerosy | I looked at my /etc/asterisk/voicemail.conf and it does not have format=x |
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| 17:26.41 | catirerosy | this is what my voicemail.conf file shows: |
| 17:26.43 | pabelanger | catirerosy: well, it is set someplace. Maybe an #included file. What version of asterisk are you using? |
| 17:26.48 | catirerosy | ;! |
| 17:26.49 | catirerosy | ;! Automatically generated configuration file |
| 17:26.49 | catirerosy | ;! Filename: voicemail.conf (/etc/asterisk/voicemail.conf) |
| 17:26.49 | catirerosy | ;! Generator: AppVoicemail |
| 17:26.49 | catirerosy | ;! Creation Date: Wed Jan 18 14:21:29 2012 |
| 17:26.49 | catirerosy | ;! |
| 17:26.51 | catirerosy | [general] |
| 17:26.53 | catirerosy | #include "vm_email.inc" |
| 17:26.55 | catirerosy | #include "vm_general.inc" |
| 17:26.57 | catirerosy | envelope = undefined |
| 17:26.59 | catirerosy | forcegreetings = undefined |
| 17:27.03 | catirerosy | forcename = undefined |
| 17:27.05 | catirerosy | mailcmd = /usr/sbin/sendmp3voicemail.pl |
| 17:27.06 | *** kick/#asterisk [catirerosy!~pabelange@asterisk/contributor-and-bug-marshal/pabelanger] by pabelanger (pastebin) |
| 17:27.17 | jamko | lol |
| 17:27.25 | *** join/#asterisk catirerosy (~elanz@75-162-196-226.slkc.qwest.net) |
| 17:27.27 | catirerosy | authpassword = xxxx |
| 17:27.29 | catirerosy | I am using 1.8 |
| 17:27.50 | catirerosy | Asterisk 1.8.11.0 |
| 17:27.52 | pabelanger | catirerosy: right, you are also using freepbx. |
| 17:27.57 | pabelanger | !freepbx |
| 17:28.03 | catirerosy | yes |
| 17:28.04 | pabelanger | ~freepbx |
| 17:28.04 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 17:28.11 | [TK]D-Fender | That doesn't look like FreePBX |
| 17:28.17 | catirerosy | I think so but not sure |
| 17:28.46 | catirerosy | wait I don't think I am using freepbx |
| 17:29.53 | pabelanger | Well, it is going to difficult to help with your configuration issues if you are using some other product to configure asterisk. |
| 17:30.04 | [TK]D-Fender | catirerosy: You are using some sort of 3rd part GUI that is doing your dirty work for you |
| 17:32.10 | catirerosy | this is more complicated than I thought :-) |
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| 17:32.11 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 17:33.47 | catirerosy | I found where the config is |
| 17:34.08 | catirerosy | and it says format wav49|wav |
| 17:34.46 | [TK]D-Fender | catirerosy: * has no part in your MP3 conversion or the mailout at that point. Your GUI's custom scripts are doing all of that.... |
| 17:34.48 | catirerosy | how do I disable it from creating the capitalize WAV files |
| 17:34.53 | [TK]D-Fender | catirerosy: and we can't help you with those |
| 17:36.19 | catirerosy | digging through I found out the admin copied a perl script from this site: http://www.voip-info.org/wiki/view/Asterisk+Voicemail and placed it under /usr/sbin and its reference by the voicemail.conf file with a variable mailcmd= |
| 17:36.33 | catirerosy | I think thats the script that is doing the conversion |
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| 19:37.39 | weinerk | Hi. Question - given combined codec capabilities between us and peer is 0x10c (ulaw|alaw|g729) |
| 19:37.40 | weinerk | What codec will be selected for the channel? Thanks |
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| 19:45.57 | p3nguin | weinerk: I would expect ulaw to be used. |
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| 20:04.16 | weinerk | p3nguin: thanks! |
| 20:04.21 | weinerk | yes that is what I see actually happen. But wanted to know what is the mechanizm for selection. |
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| 20:07.28 | carrar | Order selection in sip.conf |
| 20:11.07 | weinerk | carrar: thanks! - do you mean the order in which they appear in allow is what determines priority? |
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| 20:28.12 | carrar | yes |
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| 20:29.10 | carrar | read the example sip.conf |
| 20:29.14 | nadya_81 | Something is wrong in my setup. Is there a way I can verify that my Asterisk installation is correct, and it's just a connectivity issue i'm having ? |
| 20:29.45 | carrar | Install & compile from source so you know you are working from a sain start |
| 20:30.28 | nadya_81 | did that :) actually a console gives me "Console call has been answered" |
| 20:30.36 | carrar | ensure the file you download matches the MD5 checksome |
| 20:30.40 | nadya_81 | sorry i meant, a console call |
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| 21:05.01 | nadya_81 | anyone knows why my installation is not listening on port 5060? (i have a port=5060 in sip.conf) |
| 21:06.17 | carrar | netstat -l | grep 5060 |
| 21:06.22 | carrar | whats the output of that |
| 21:06.58 | carrar | it should also be 'bindport=5060' |
| 21:07.01 | carrar | FTY |
| 21:09.42 | carrar | or 'udpbindaddr=0.0.0.0:5060' |
| 21:10.23 | nadya_81 | thanks i'll try this out |
| 21:10.29 | carrar | <carrar> netstat -l | grep 5060 |
| 21:10.33 | carrar | Whats the output of that |
| 21:10.49 | nadya_81 | funny is that I just configured it on a test server VM and it worked fine... when things go on the real thing they give ma a hard time |
| 21:10.59 | carrar | Whats the output of that |
| 21:11.12 | nadya_81 | nothing shows. however on my other test installation that works netstat -tulpn | 5060 works fine. |
| 21:11.19 | carrar | k |
| 21:11.25 | nadya_81 | i mean shows the asterisk entry |
| 21:11.47 | carrar | netstat -l | grep 5060 |
| 21:11.48 | carrar | udp 0 0 *:5060 *:* |
| 21:11.52 | carrar | thats the expected output |
| 21:11.57 | carrar | that means it's working |
| 21:12.02 | carrar | and you probably have a filtering issue |
| 21:12.07 | carrar | iptables etc... |
| 21:15.08 | carrar | You live in malta? |
| 21:15.31 | nadya_81 | yea... |
| 21:15.36 | carrar | nice Island |
| 21:15.48 | nadya_81 | been here ? |
| 21:15.56 | carrar | Never |
| 21:16.28 | nadya_81 | it has got some nice parts |
| 21:16.40 | carrar | It's a resort island? |
| 21:16.45 | carrar | like Hawaii? |
| 21:17.29 | eZz | is it possible to specify >1 of country codes in section general in /etc/asterisk/indications.conf ? |
| 21:17.38 | nadya_81 | nope, i'd say a good mix of things. history, good food, nightlife and beaches |
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| 21:18.15 | nadya_81 | you're not going to find the 'luxury' and pure nature of hawaii over here |
| 21:18.25 | carrar | heh |
| 21:20.20 | carrar | Oh I see the Radison resport |
| 21:20.23 | carrar | resort |
| 21:20.27 | carrar | looks fancy |
| 21:20.42 | eZz | is it possible to specify >1 of country codes in section general in /etc/asterisk/indications.conf ? |
| 21:21.12 | carrar | Ghajn Tuffieha Bay |
| 21:22.25 | nadya_81 | yeah exactly. that is really nice, my favorite beach |
| 21:23.37 | *** join/#asterisk AndChat|296121 (~AndChat29@mail.jaguardata.com) |
| 21:25.20 | carrar | eZz, why would you do that? |
| 21:26.13 | carrar | Just put settings in the global and define 1 |
| 21:26.28 | carrar | crazy though |
| 21:27.01 | carrar | I've not tried |
| 21:29.12 | *** join/#asterisk danfromuk (~IceChat77@2.27.26.23) |
| 21:29.32 | danfromuk | Hi, Im moving from 1.4 to 1.8 and having a problem executing AGIs. |
| 21:29.34 | carrar | Majorca is as close to your island as I've been :) |
| 21:29.50 | nadya_81 | where do you come from? european? |
| 21:29.53 | danfromuk | I'm getting the following error: pbx_extension_helper: No application 'AGI,/path/to/pre_call_check.php' for extension .... |
| 21:30.01 | carrar | Seattle |
| 21:30.09 | carrar | USA |
| 21:30.10 | danfromuk | I've run 'core show applications' and AGI is listed. |
| 21:30.17 | carrar | (Bellevue) |
| 21:30.44 | carrar | I've in Berlin a year ages ago |
| 21:30.44 | carrar | Lived |
| 21:33.05 | carrar | danfromuk, your path is wrong |
| 21:33.19 | carrar | or you put your agi file in the wrong directory |
| 21:34.57 | carrar | module show like res_agi.so |
| 21:35.33 | p3nguin | If the app usage shows up, doesn't that imply the module is loaded? |
| 21:36.57 | carrar | danfromuk, binpaste your dialplan where you are using it |
| 21:37.05 | carrar | ~binpaste |
| 21:37.12 | danfromuk | i checked the path, but i'll double check. |
| 21:37.24 | carrar | ~paste |
| 21:37.24 | infobot | extra, extra, read all about it, paste is http://pastebin.org/ or http://bin.cakephp.org/ or http://pastebin.ca/ |
| 21:37.27 | p3nguin | ~pb |
| 21:37.27 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 21:37.57 | danfromuk | The path is correct. |
| 21:38.03 | danfromuk | 1sec. i'll pb it |
| 21:38.13 | carrar | How did you verify that? |
| 21:38.46 | carrar | grep agi-bin /etc/asterisk/asterisk.conf |
| 21:39.51 | danfromuk | http://pastebin.com/LZCmp8ev |
| 21:40.24 | danfromuk | the php script is in the location listed in the diaplan code. |
| 21:40.25 | p3nguin | AGI() |
| 21:40.29 | p3nguin | You've got AGI, |
| 21:40.33 | carrar | heh |
| 21:40.49 | p3nguin | AGI(pre_call_check.php) |
| 21:41.02 | carrar | core show application AGI |
| 21:41.23 | carrar | almost there :) |
| 21:41.26 | p3nguin | And put it in your agi-bin path instead of the scripts path. |
| 21:41.30 | carrar | yes |
| 21:41.37 | danfromuk | Ah. I didnt see that change listed in the UPGRADE documents |
| 21:41.38 | danfromuk | ok |
| 21:41.42 | danfromuk | i'll give that a go |
| 21:41.45 | carrar | HAVING FUN? :) |
| 21:42.20 | carrar | guess not |
| 21:42.22 | carrar | heh |
| 21:45.40 | danfromuk | perfect. thanks! |
| 21:56.52 | *** join/#asterisk fr0ggie (~irc@unaffiliated/nn) |
| 22:02.25 | fr0ggie | Im having an issue when i try to call one of my extensions (well any of them it seems) -- WARNING[21264][C-00000009]: app_dial.c:2333 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 22:05.44 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 22:07.20 | afink | fr0ggie: sounds like your extensions aren't registered, does "sip show peers" show your extensions? |
| 22:07.54 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 22:09.32 | fr0ggie | aha.. now why are my phones falling off...hmm |
| 22:09.47 | WIMPy | gravity |
| 22:09.55 | fr0ggie | phone shows Registered at 1000-- forcibly re-registering it makes it reachable |
| 22:10.15 | afink | I am having a one way audio issue with a remote sip extension, I can receive incoming calls however I am unable to get audio on outgoing calls. I have configured both firewalls to port forward 10000-20000 and 5060. |
| 22:10.57 | afink | fr0ggie: this is a shot in the dark, but I had the same problem. Is your firewall open for SIP DoS attacks? That is what kept knocking my extensions offline. |
| 22:11.02 | WIMPy | afink: Are you sure the phone uses an UDP port in that range? |
| 22:11.17 | WIMPy | Looks like you both have nat issues. |
| 22:11.20 | fr0ggie | afink: its not reachable outside my LAN |
| 22:11.26 | WIMPy | That's perfectely normal. |
| 22:11.34 | fr0ggie | no NAT either |
| 22:11.44 | p3nguin | afink: "sip show peers" shows peers, not extensions. |
| 22:11.58 | WIMPy | Or just firewall. |
| 22:12.54 | fr0ggie | WIMPy: no firewall even, * runs on my laptop which has an empty iptables. Phones are nearby on same subnet (not passing through the router or any of the bridges on the network) |
| 22:13.20 | fr0ggie | i havent attempted to get NAT (for outside the network) working yet even |
| 22:18.27 | afink | WIMPy: yes, yealink says RTP ports are from 11780-11800 |
| 22:20.48 | afink | that is what you were asking correct? |
| 22:21.08 | WIMPy | yes |
| 22:21.42 | afink | i also noticed when configuring the phone all but one of the codecs were disabled so I re-enabled all of the codecs also |
| 22:27.11 | afink | which also didn't fix it. |
| 22:33.36 | *** part/#asterisk nW44b (~Schnitzel@unaffiliated/benwa) |
| 22:37.09 | *** join/#asterisk afink (~chatzilla@ip68-13-94-224.om.om.cox.net) |
| 22:37.11 | fr0ggie | WIMPy: any ideas why my phones are losing registration? I use TCP to connect to * as it was recommended to avoid issues when I get started with accessing the * box from behind a NAT (on my 3g) |
| 22:39.14 | WIMPy | tcpdump/wireshark/etc are the tools of choice for networking issues. |
| 22:40.14 | p3nguin | Did you set up asterisk for tcp? |
| 22:41.08 | carrar | could have solved all your problems by just using a ipsec vpn over your 3g and leave everything be |
| 22:42.06 | nadya_81 | managed :) |
| 22:44.09 | fr0ggie | p3nguin: yae-- i can connect i just lose registry sometimes- Im wondering if its a bug in the android SIP ua or if its maybe that i need to disable qualify since im using tcp |
| 22:44.42 | fr0ggie | carrar: i dont have room in router's flash for ipsec and l2tpd, unfortunately |
| 22:44.58 | fr0ggie | Had to give them up to make room for freeradius |
| 22:45.44 | WIMPy | qualify works with tcp as well. |
| 22:45.48 | carrar | fr0ggie, don't need a router |
| 22:48.04 | carrar | racoon is your friend! |
| 22:48.47 | fr0ggie | carrar: i'd have to have somewhere for the VPN client to connect (thus my openwrt router) |
| 22:49.00 | fr0ggie | asterisk runs on a netbook i use as the house server |
| 22:49.07 | carrar | same server Asterisk is running on |
| 22:49.28 | carrar | openwrt might already support that |
| 22:49.33 | carrar | (I don't use that) |
| 22:50.19 | fr0ggie | might be able to do some iptables-fu to pass the ipsec traffic to the * box but in the end im thinking asterisk will end up on the openwrt box again (it works well there using the config ive got on the netbook-server) |
| 22:50.34 | fr0ggie | i still suspect its an android bug |
| 22:58.52 | ChannelZ | wouldn't be the first |
| 23:01.24 | fr0ggie | and now they've all dropped off again in sip show peers |
| 23:01.43 | fr0ggie | fr0ggie/fr0ggie (Unspecified) D 0 UNKNOWN |
| 23:04.42 | p3nguin | They're on the same switch? |
| 23:06.03 | p3nguin | I've been having some issues with a couple ATAs on the same switch as the asterisk computer behaving similar to that. At least mine aren't going completely away. |
| 23:06.43 | fr0ggie | they're wireless on the same AP as * |
| 23:07.00 | fr0ggie | phone still shows it's registered even when * doesnt think they are anymore |
| 23:07.49 | WIMPy | Networking disabled to save power? |
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| 23:44.45 | logicwrath | anyone using ActivaTSP on a Windows 2008 Server? |
| 23:56.27 | *** join/#asterisk nix8n82 (~nate@24.143.10.144) |