| 00:04.45 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 00:04.54 | *** join/#asterisk wdoekes (~walter@wjd.osso.nl) |
| 00:10.16 | navaismo | I just upgrade to the lates branch of asterisk 12 and now im receiving-->"config_options.c:703 aco_process_var: Could not find option suitable for category '5000' named 'dtmfmode' at line 0 of " |
| 00:16.44 | Penguin | line 0 of what? |
| 00:26.30 | navaismo | im wondering the same |
| 00:26.42 | navaismo | the log end in that word |
| 00:26.45 | Penguin | That was the end of the message? |
| 00:26.48 | navaismo | yes |
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| 00:28.19 | outtolunc | an include with a non-existent file? ;) |
| 00:29.30 | navaismo | <PROTECTED> |
| 00:29.41 | file | if using pjsip some of the options changed names to unify them |
| 00:30.52 | navaismo | weird the table looks the same as the previous |
| 00:31.37 | file | dtmfmode is dtmf_mode now I believe |
| 00:31.38 | navaismo | argh dtmf_mode instead dtmfmode |
| 00:31.50 | navaismo | yup |
| 00:32.50 | navaismo | ugh many names add a underscore |
| 00:34.16 | navaismo | s/a/an/ |
| 00:34.29 | navaismo | pff |
| 00:35.03 | Penguin | lol |
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| 00:41.42 | karl-s | i'm having trouble understanding the correlation between 'allow' statements in SIP.conf and SDP. does ordering of the allow statements really matter? |
| 00:42.17 | Penguin | The order is supposed to show preference. I recommend allowing only the codec you want to use. |
| 00:43.05 | karl-s | so if peer 1 calls peer 2, both have the allow order of (g729,alaw,ulaw), when asterisk sends the invite for peer 2, only g729 is in the SDP??? |
| 00:43.08 | karl-s | is that normal? |
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| 00:43.38 | Penguin | If you allow all of them, the SDP should show all of them are available. |
| 00:43.43 | karl-s | right! |
| 00:43.46 | karl-s | thats what i thought |
| 00:44.04 | karl-s | but for some reason only g.729 shows up |
| 00:44.14 | Penguin | Are you using one allow line with a comma-separated list, or three allow lines with one codec per line? |
| 00:44.27 | karl-s | line by line |
| 00:44.50 | karl-s | if g.729 is first, only g.729 shows up |
| 00:45.03 | karl-s | if alaw or ulaw is first, only alaw or ulaw shows up |
| 00:46.40 | navaismo | now-->Error parsing call_group= at line 0 of |
| 00:49.09 | navaismo | hmm iset to NULL resolve the error |
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| 00:51.36 | navaismo | weee the websocket still killing asterisk |
| 00:52.05 | navaismo | im going to try the beta2 package |
| 00:58.01 | file | it has not been fixedvyet |
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| 01:08.28 | navaismo | indeed :'( im wondering why other people can use it with asterisk 12 is this a thing of using a db? |
| 01:17.34 | file | well, you can still use chan_sip... where it continues to work |
| 01:20.53 | Penguin | Is chan_sip actively being broken with new asterisk versions? |
| 01:21.52 | *** part/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
| 01:22.02 | WIMPy | It did break for me. |
| 01:22.40 | WIMPy | Got one-way-audio on calls where Asterisk had to transcode. |
| 01:23.09 | WIMPy | Or that's what I think was what' was going on. |
| 01:24.01 | navaismo | is there a log of the ari connections in the asterisk cli like manager does? |
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| 01:42.12 | navaismo | ok this is strange the websocket connection coming from wscat.py script works fine but from jssip/sipml5 just kill asterisk |
| 01:42.39 | file | its not strange, if you know what is happening |
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| 01:43.18 | navaismo | obviously not |
| 01:44.15 | file | the SIP traffic over the websocket causes it, not a generic websocket connection |
| 01:44.55 | file | specifically the code that determines what transport is in use does not work on the websocket transport |
| 01:45.09 | navaismo | thanks |
| 01:47.25 | navaismo | i need to read a lot really, this new sorcery about Stasis and ari drive me crazy! So from a websocket i create a new app and then it can be accesed from the dialplan aaaaaaa |
| 01:47.32 | navaismo | go for a beer |
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| 02:02.58 | hebber | Have anyone bought Digium g729 licenses - how do they provide the license keys - only got a packing slip and a invoice |
| 02:04.25 | Penguin | It's been a while, but it seems like it comes in email. |
| 02:05.54 | Penguin | I'm also curious about a packing slip for an intangible product. |
| 02:06.30 | hebber | Penguin: me too - it even says NOSHIP under shipping method, must be an error - thanks anyway |
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| 05:07.54 | road33 | I have someone on my server trying to dial international numbers, there are no sip peers, other than the expected ones, how are they doing this? |
| 05:08.46 | Penguin | You allow anonymous calls. |
| 05:08.56 | road33 | not on purpos |
| 05:09.12 | Penguin | sip.conf allowguest |
| 05:10.05 | road33 | hummm, unset, default is yes it seems |
| 05:10.21 | Penguin | Check the sample file. |
| 05:11.03 | road33 | ok, that maybe it |
| 05:11.46 | Penguin | Which asterisk branch are you using? |
| 05:13.05 | road33 | humm, not finding out how to figure that out, its ubuntu 13.10 |
| 05:13.20 | Penguin | Which asterisk branch are you using? |
| 05:13.37 | Penguin | You're saying you don't know how to find out the asterisk version? |
| 05:13.40 | road33 | yes |
| 05:13.43 | road33 | branch |
| 05:13.50 | Penguin | When you connect to the CLI, what does it say the version is? |
| 05:14.54 | road33 | Asterisk 1.8.10.1~dfsg-1ubuntu1 |
| 05:15.21 | Penguin | I was thinking allowguest was default no in 1.8 branch. I'll look in my sample file. |
| 05:15.45 | road33 | the sample file documentation says, default is yes, but it could be an old config file. |
| 05:16.00 | Penguin | ;allowguest=no ; Allow or reject guest calls (default is yes) |
| 05:16.13 | road33 | I changed this to yes |
| 05:16.19 | road33 | I mean no, |
| 05:16.31 | Penguin | If you set allowguest=yes, that is the same as not setting it. |
| 05:16.42 | road33 | yes, sorry its a bit late here |
| 05:17.11 | Penguin | Are you using fail2ban to block attacks and dial attempts? |
| 05:17.23 | road33 | I just added that prior to joining channel |
| 05:17.28 | WIMPy | Hmm. is amrnb required and not optional? Or am I missing something? |
| 05:18.04 | Penguin | With it set to not allow guests, you will probably see messages about fake auth. The source IP address will not be shown, and fail2ban will not be able to block the IP address. |
| 05:18.38 | Penguin | In asterisk 10 and up, there was a new logging facility provided which logs only those addresses, but we don't have it in 1.8... |
| 05:18.39 | road33 | yes, I already see the fake auth, is there a way around this? |
| 05:19.44 | road33 | I can update, maybe tomarrow |
| 05:19.47 | Penguin | So in 1.8, I leave allowguest set to yes and create a type of catch-all extension that all those "dialers" will hit. The extension uses the Log() application to write a string which I then use in fail2ban to block them. |
| 05:20.08 | road33 | well that makes a lot of sense |
| 05:20.13 | Penguin | If you're going to update from 1.8, go to 11. |
| 05:20.27 | road33 | ok, I will do that tommarow |
| 05:20.52 | road33 | I have internaltional calling blocked, so I shouldn't get charged |
| 05:20.59 | Penguin | If you are interested in doing it how I do it and need help, I'll tell you step by step how I set up mine. |
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| 05:21.38 | road33 | well thats very helpful, but I think I will wait on that and do some research |
| 05:21.48 | road33 | and try to make changes tomarrow |
| 05:21.51 | Penguin | Actually, it's not even really a bunch of steps. |
| 05:21.59 | road33 | ok, then I am game |
| 05:22.29 | Penguin | With allowguest=yes, the context that is set in the general section is the context your anonymous calls go to. |
| 05:22.49 | Penguin | I set context=unauthenticated in the sip.conf general. |
| 05:23.48 | road33 | got ya |
| 05:24.29 | Penguin | In the unauthenticated context, I have a couple extensions... |
| 05:25.01 | road33 | what do they hit a lot? |
| 05:25.23 | Penguin | http://pastebin.com/pr3HeEyZ |
| 05:25.57 | Penguin | Then in the fail2ban asterisk.conf, I add another regex line NOTICE%(__pid_re)s .*: <HOST> is attempting to make unauthorized calls |
| 05:26.46 | Penguin | They usually dial 011.... and 9011.... and 99011.... |
| 05:27.02 | Penguin | And sometimes they throw a plus sign on the front of those extensions. |
| 05:27.36 | road33 | ok, I made the changes |
| 05:27.45 | road33 | thanks for your help, |
| 05:28.02 | road33 | I was very upset |
| 05:28.20 | Penguin | As long as fail2ban is reading the log file and the NOTICEs are showing up, you'll see the IPs getting blocked in iptables. |
| 05:28.54 | road33 | yes, I am waiting for that now, seems they may have already moved along |
| 05:29.12 | Penguin | Be sure to restart fail2ban, sip reload and dialplan reload. |
| 05:29.56 | Penguin | Lately, the attempts have been spaced far apart, so the findtime in fail2ban needs to be extended to at least a couple hours. |
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| 05:30.32 | road33 | I have failtime at 3600 |
| 05:30.49 | Penguin | 30 minutes? |
| 05:31.15 | Penguin | I'd up it to at least two hours. |
| 05:31.24 | Penguin | 7200? |
| 05:31.43 | Penguin | having trouble with late-night math |
| 05:32.02 | Penguin | infobot: convert 2 hours to seconds |
| 05:32.24 | Penguin | So your 3600 is one hour, then. I'd increase it. |
| 05:32.33 | road33 | got yea, reset it 7200 |
| 05:33.00 | road33 | ok, good got some action on iptables now |
| 05:33.31 | Penguin | Yep, when you change the find time, fail2ban will go back in the log file and ban retroactively. |
| 05:34.30 | road33 | so I need to change allowguest=yes again? seems the fails were from before |
| 05:34.36 | Penguin | Yes. |
| 05:34.53 | Penguin | With allowguest set to no, you'll get the fake auth messages. |
| 05:34.56 | ChannelZ | (do you need/want anonymous calls?) |
| 05:35.14 | Penguin | You get the fake auth messages in lieu of the offenders' addresses. |
| 05:35.38 | Penguin | So you have to allow anonymous calls for the trap to work. |
| 05:36.20 | road33 | seems like we should just share offending ip address and get a common ban list |
| 05:36.27 | Penguin | If you set allowguest to no, the only IP address that shows up is your own. You don't really want to ban yourself. |
| 05:36.40 | road33 | agree :) |
| 05:37.00 | Penguin | I have a huge list, both addresses and networks. |
| 05:38.09 | Penguin | I'm not very forgiving, though. |
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| 05:39.15 | road33 | thank you for your help |
| 05:39.33 | road33 | I will have to wait it seems to catch someone |
| 05:39.50 | Penguin | No problem. Let me know how it serves your needs. It works great for me. |
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| 06:41.05 | b7 | Hello everyone. I have a problem with the attended call transfer. The transfer itself works fine, the problem is the displayed number. I have a SPA303/921 sip phones with the "xfer" softkey, so when a user going to transfer an incoming call to another user he press the "xfer" key and the phone makes the second call to another local user and then do the transfer when the second user hook off. But another user see a number of the first user, not the incoming numbe |
| 06:41.06 | b7 | r. I need to notify the second user that this is a transfered incoming call, not a local call. But have no idea how. |
| 06:43.42 | b7 | I tried to setup sendrpid and trustrpid in sip.conf, but the best I could got is a number replacing with rpid_update after the second user accept the call. But I need to notify the user before he do it. |
| 06:45.09 | b7 | Please can anybody help me to solve the problem? |
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| 07:22.37 | ChannelZ | That's how attended transfers work. You, the transferer, are making a new call to the person you're transferring to. |
| 07:23.09 | ChannelZ | Unless you need to talk to the person you're transferring to first, do a blind transfer.. which unfortunately on the SPAs is on the second page of the menu, you have to scroll to the right |
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| 08:41.16 | b7 | But before we had the SPA9000 instead of Asterisk and in the case of a call transfer the displayed number was a number of incoming call. There should be some possibility to do that, I sure. |
| 08:46.19 | ChannelZ | IE Call Manager? Totally different beast |
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| 08:54.48 | b7 | Hmm... so is there no way to do the trick with asterisk? We need to add a 'Remote-Party-ID' header with a proper incoming number to the second call, but the problem is how to detect that the phone going to do the transfert. The second call is separate from the first and pretty usual until the transfer happens... |
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| 08:56.07 | b7 | Maybe some phone's setting that lets to setup the variables or something? |
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| 10:59.51 | morcao | good morning |
| 10:59.56 | morcao | I have one problem, I can't receive call to my softphone but i can connect from this |
| 11:00.10 | morcao | and other problem is the call is ended after the 30 seconds |
| 11:00.37 | wdoekes | end after 30 seconds, sounds like the SIP ACK not reaching the right destination |
| 11:00.40 | morcao | someone passed for this ? |
| 11:01.07 | wdoekes | morcao: begin by debugging the second problem. sip set debug on |
| 11:03.34 | morcao | Really destroying SIP dialog '3588cd5-29a7799e-52947fc2@'MY IP' Method: REGISTER |
| 11:03.38 | morcao | I received this |
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| 11:07.42 | morcao | call between zoipers is working |
| 11:07.59 | morcao | but call's between zoiper and X-litle not Working :/ |
| 11:08.15 | tomodachi | morcao: codec problem? |
| 11:08.25 | tomodachi | i know that x-lite does not always honour codec priority |
| 11:11.43 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 11:12.30 | morcao | I've had this problem in zoiper but I solved this |
| 11:12.51 | weinerk | Please help. meetme + ast1.4 + SIP + agi - need moderator to have custom control by dtmf |
| 11:14.00 | morcao | I think this is one problem of codec but is strange, X-litle didn't received the call |
| 11:15.37 | morcao | weinerk explain our problem |
| 11:17.15 | weinerk | marcao: 1.thanks 2.details - I have ast ver 1.4, using meetme for conferencing, but I need a function that is not part of meetme by default |
| 11:18.05 | weinerk | I want to have a moderator who can click some key and mute all or unmute all |
| 11:20.57 | weinerk | morcao: any advice? |
| 11:23.33 | morcao | You have to do a script to go do it, I don't know this functionality |
| 11:25.12 | morcao | you can do so from the application side easily |
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| 12:40.29 | ipalmer | hello all, I'm trying to get an android phone, using Zoiper to register with Asterisk over 4g, the trouble is when I register, which is successfull, the asterisk realtime database shows the ipaddr column as an internal ip address in the 10 range, why is this |
| 12:41.50 | eirirs | ipalmer: are your phone on wlan too? |
| 12:43.21 | ipalmer | eirirs: no, the phone is using 4f and I have a 4g router which it is configured to use |
| 12:43.26 | ipalmer | 4g |
| 12:44.58 | ipalmer | eirirs: the thing is I can make inbound and outbound calls ok, but I have no sound on the internal phone when talking into the android handset, but I have sound on the andoind when talking into the internal handset |
| 12:45.59 | eirirs | sounds like nat issue |
| 12:47.02 | ipalmer | I have configured asterisk to use ports 10000 - 10010 and port forwarded each of these ports |
| 12:50.46 | eirirs | set localnet and externip ? |
| 12:51.33 | eirirs | ipalmer: http://www.voip-info.org/wiki/view/Asterisk+SIP+externip |
| 12:51.38 | ipalmer | yep |
| 12:51.55 | ipalmer | though it is called externaddr now? |
| 12:52.25 | eirirs | dont remember, just try externip and see what happens |
| 13:03.38 | ipalmer | eirirs: changed externaddr to externip but this has made no difference |
| 13:09.05 | eirirs | have you any sip alg or any sip options enabled in router ? |
| 13:09.30 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 13:10.14 | ipalmer | SIP alg is disabled on the router |
| 13:11.27 | eirirs | did you see the port config for 1000-10010 at zoiper? |
| 13:11.31 | eirirs | err 10000-10010 |
| 13:12.21 | ipalmer | the 10000-10010 is the rtp ports in use, I can't find anywhere to configure the rtp ports in zoiper |
| 13:13.35 | eirirs | there should be |
| 13:16.26 | *** join/#asterisk anonymouz666 (~anonymouz@187.76.181.102) |
| 13:16.31 | ipalmer | ok, I have been looking for this setting all day but can't find it |
| 13:17.28 | eirirs | ipalmer: zoiper - settings - accounts - sip account - network settings |
| 13:18.26 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 13:19.22 | ipalmer | thanks, the only fields in there though are registration expiry time, transport type, nat use stun, stun server, stun port, stun refresh, use rport for signalling and use rport for media |
| 13:20.16 | eirirs | ipalmer: could it be of some help? http://stackoverflow.com/questions/14300979/why-asterisk-not-properly-working-with-android-sip-client |
| 13:20.59 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 13:21.15 | eXcAliBuR | what file or files do I need to provision a digium D50 using a URL ? |
| 13:25.38 | b7 | Hmm, I need to play a message in BackGround() and do a Dial() at the same time. How can I do that? |
| 13:29.16 | *** join/#asterisk Draecos (~Draecos@106-68-225-37.dyn.iinet.net.au) |
| 13:30.27 | LooserOuting | maybe a custom ringback tone ? |
| 13:31.15 | LooserOuting | why not first play the message and then dial ? |
| 13:34.01 | *** join/#asterisk michael_work (~michael@bzq-112-168-31-118.red.bezeqint.net) |
| 13:35.08 | b7 | Well, this is a voice message for an office off-hours but the phones must ring in a case if somebody still in office... |
| 13:36.45 | eXcAliBuR | ring first then |
| 13:36.55 | eXcAliBuR | message when not answered |
| 13:36.57 | eXcAliBuR | :) |
| 13:37.33 | *** join/#asterisk stanman246 (~chatzilla@a80-100-37-110.adsl.xs4all.nl) |
| 13:37.44 | LooserOuting | I would go with custom ringback tone |
| 13:38.16 | LooserOuting | instead of sending ringbacktone play an audiofile |
| 13:40.08 | stanman246 | hi in here, i've got raspbx up and running with an iax2 trunk, proud as hell :D Now I'd like my polycom phones to use a centralized addressbook. I've been reading about superfecta and the phonebook module, but am not sure where to start. Anyone have a similar setup? (asterisk, freebpx, phonebook module) |
| 13:40.44 | b7 | Boss wants to do that at the same time, if possible. Custom ringback seems to be the solution, thank you. ;) |
| 13:41.21 | LooserOuting | b7: just use m in the dial i.e. exten => ....37,1,Dial(Sip/37,20|m(nameofmusicfile)) |
| 13:41.42 | LooserOuting | or something like that :) |
| 13:42.26 | b7 | LooserOuting, going to try this. |
| 13:47.59 | eXcAliBuR | can I trade bitcoin for support? |
| 13:50.11 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:02.41 | *** join/#asterisk saliak (~saliak@ip72-195-155-142.ri.ri.cox.net) |
| 14:03.39 | *** join/#asterisk redotis (~redotis@nc-76-5-176-123.dhcp.embarqhsd.net) |
| 14:03.49 | saliak | trying to debug BLF issues. Is there a way to display explicitly who the watchers are for a list of extensions? I can see how many with core show hints, but not who… right now it's only listing one (for one of my extensions), rather than two (the two extensions that are watching) |
| 14:04.41 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 14:05.44 | anonymouz666 | it is possible. you must look for SUBSCRIBE method using SIP debug |
| 14:06.01 | saliak | ok |
| 14:07.04 | *** part/#asterisk anonymouz666 (~anonymouz@187.76.181.102) |
| 14:08.14 | ipalmer | asterisk 11.5 sip realtime nat settings, it says NAT = yes is deprecated and should use force_rport, comedia, will this just be a case of me changing the enum on the database or is there more to it? |
| 14:10.16 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 14:11.10 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:15.40 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 14:22.05 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:22.05 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:25.20 | saliak | Also, I have a phone that registers successfully and I can call it, however, they can't hear anything (handset or speakerphone). really weird. Have tried changing volume, calling #s in and out of our office with similar results. any way this would be an asterisk thing? |
| 14:26.02 | eXcAliBuR | i had that problem, needed a firmware update |
| 14:26.19 | eXcAliBuR | was the polycom IP something |
| 14:26.51 | [TK]D-Fender | saliak: Details + Debug |
| 14:27.40 | saliak | [TK]D-Fender: yeah. that's all i got for now, just seeing if anyone had heard of something similar. I'm going to try the firmware update first then go down the sip debug road. |
| 14:29.41 | *** join/#asterisk kannan (~chatzilla@112.79.46.104) |
| 14:31.23 | [TK]D-Fender | saliak: SIP DEBUG is the FIRST road to take |
| 14:32.00 | [TK]D-Fender | saliak: Assuming firmware as being buggy is pretty much the last thing to look at. |
| 14:32.03 | kannan | hi, if i set cdr-adaptive_odbc to wrtie cdrs to 2 different DB servers, apart from localhost, and should any one cdr odbc write fail, will that terminate call executions in suceeding h priorities? If we have h extension for the context, will the cdr be the last item executed after all h priorites? |
| 14:32.45 | *** join/#asterisk calum_ (~calum_@cpc4-harg5-2-0-cust371.7-1.cable.virginm.net) |
| 14:33.20 | kannan | also, in an agi arg, a value that is comma delimited, like firstname,lastname gets treated as two separate arguments, even if i double quote it. is there any way to get around that, or is it something to be handled always in teh agi only and concatenate them there? |
| 14:35.45 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-zsejwqfryzrryerv) |
| 14:35.45 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:36.33 | *** join/#asterisk Takapa (vegard@junior.svanberg.no) |
| 14:38.58 | [TK]D-Fender | kannan: just leave it in a variable and let the AGI read it instead of passing it as an arg |
| 14:40.14 | kannan | oh ok, thats it. it didnt strike me at all |
| 14:41.02 | kannan | is the arg character count maximum allowed 255 ? |
| 14:41.31 | [TK]D-Fender | kannan: not sure.. |
| 14:42.04 | kannan | ok but this way of reading the channel var in agi ,that doesnt matter |
| 14:42.25 | kannan | thanks, any advise for the cdr question qbove? |
| 14:42.28 | kannan | above |
| 14:44.35 | [TK]D-Fender | kannan: I don't recall CDR updating for "h" at all... |
| 14:45.28 | *** join/#asterisk Joel_re (~jr@115.69.254.4) |
| 14:45.53 | Joel_re | hey, what is a PRI like GSM connection called? |
| 14:45.56 | kannan | [TK]D-Fender , i meant to ask what happens if one ODBC connectivity fails, say for network outage reason for example. will it casue call to fall through and not execute any h,n, applciation |
| 14:46.09 | Joel_re | can that be terminated at a customer premises box? |
| 14:46.27 | [TK]D-Fender | kannan: Shouldn't. CDR is "best efforts" |
| 14:46.53 | [TK]D-Fender | Joel_re: huh? |
| 14:46.56 | kannan | [TK]D-Fender , thanks. |
| 14:46.59 | Greenlight | You can, and likely should at large volumes, run CDR in batch mode |
| 14:47.15 | Greenlight | This way you guarentee the abstration |
| 14:47.35 | [TK]D-Fender | Joel_re: PRI is PRI .... GSM is either in reference to the codec, or to the cellular wireless standard... neither has anything to do with the other... |
| 14:47.46 | kannan | Greenlight , new to me , whats large volume, we dont have more than 4 concurrent T1s max |
| 14:47.48 | Greenlight | From my limited testing with it, when DB connections go down, it caches till they are back up |
| 14:47.54 | Joel_re | [TK]D-Fender: right, I worded it wrongly |
| 14:48.11 | [TK]D-Fender | Joel_re: Try again... |
| 14:48.13 | eXcAliBuR | anyone wanna look at my res_digium_phone.conf http://pastebin.com/AaBjS5XR ?? |
| 14:48.24 | Joel_re | is there anything like a cellular PRI line, basically a cellular number that will accept multiple callers |
| 14:48.25 | eXcAliBuR | maybe can see something i'm missing |
| 14:48.34 | Joel_re | or does it matter if its a cellular line or not |
| 14:48.44 | Joel_re | s/line// |
| 14:48.47 | kannan | Greenlight , but i am still on 1.8.24.0 version.. |
| 14:49.08 | Greenlight | I beleive batch mode was available in 1.8 |
| 14:49.42 | [TK]D-Fender | Joel_re: there is no such thing as a "cellular number" |
| 14:50.03 | Joel_re | ok so numbers can be anything, they are hooked to cellular networks |
| 14:50.04 | *** join/#asterisk serafie (~erin@nat/digium/x-dcjawsycvvipwrat) |
| 14:50.05 | [TK]D-Fender | Joel_re: leach cell call occupies a channel to a device. Multiple calls = multiple radios |
| 14:50.13 | kannan | Greenlight , the wiki says "Use of batch mode may result in data loss after unsafe asterisk termination, i.e., software crash, power failure, kill -9, etc." |
| 14:50.25 | Greenlight | kannan: Indeed. |
| 14:50.26 | [TK]D-Fender | Joel_re: So you want multiple Cell channels : get a multi-SIM devices |
| 14:50.45 | Joel_re | [TK]D-Fender: what if I need over 30-40 channels |
| 14:50.45 | Greenlight | I figure if you care about your CDR's you've at least got the server on UPS. |
| 14:50.57 | [TK]D-Fender | Joel_re: a TON of them then. |
| 14:50.58 | Greenlight | GSM gateway |
| 14:51.13 | Joel_re | do cellphone companies give us a number? |
| 14:51.21 | [TK]D-Fender | Joel_re: Go ask your cell co. |
| 14:51.28 | [TK]D-Fender | Joel_re: How service works is up to them |
| 14:51.30 | kannan | Greenlight , yes sure, its in a datacenter with a backup failover server too |
| 14:51.30 | Joel_re | ok, Im just asking in general |
| 14:51.46 | Joel_re | wondering if they will terminate it to my host in a datacenter |
| 14:52.27 | [TK]D-Fender | Joel_re: I doubt they will offer you any such deployment method |
| 14:52.33 | Greenlight | AT least here in the UK you can't use a "mobile" number for a landline, only way is actually using SIM card and radio (GSM gateway etc) |
| 14:52.53 | Joel_re | or maybe a multisim to sip termination |
| 14:56.20 | kannan | how can i bind multiple sip ports in sip.conf with bindport setting, is it allowed inside the context for a service provder. so i can have provider1 - 5060 , provider2 - 5072 ? |
| 14:56.44 | [TK]D-Fender | kannan: you can't |
| 14:56.50 | [TK]D-Fender | kannan: * binds one port. |
| 14:56.58 | kannan | oh |
| 14:57.23 | [TK]D-Fender | kannan: Why does provider 2 care what port YOU use? |
| 14:57.29 | kannan | i dont know |
| 14:57.54 | [TK]D-Fender | kannan: Can youshow us where they are requesting it this way? |
| 14:57.55 | kannan | they have said they need that, it wwas not registering with UN / pwd , |
| 14:58.09 | kannan | one min ple, i will put it in a pastebin |
| 14:58.35 | *** join/#asterisk calum_ (~calum_@cpc4-harg5-2-0-cust371.7-1.cable.virginm.net) |
| 15:03.00 | kannan | i am not sure how to pastebin it, it is simply consisting of two regter lines in sip.conf, for one account they said we must use port 15060, it is a singapore voip provider |
| 15:05.14 | [TK]D-Fender | kannan: I wasn't asking for your config... I was asked for some doc where thy say that YOUR listening port has to be something non-standard |
| 15:05.37 | [TK]D-Fender | kannan: You sounds like you've mixed up your understanding of what port they want you to send TO.... |
| 15:05.55 | [TK]D-Fender | kannan: that is THEIR portm not YOUR port |
| 15:06.08 | kannan | [TK]D-Fender , ok , what setting do i use for that |
| 15:06.17 | [TK]D-Fender | port= <- |
| 15:06.20 | [TK]D-Fender | in your peer |
| 15:06.30 | [TK]D-Fender | bindport is a [general] setting for YOUR server |
| 15:06.37 | kannan | [TK]D-Fender , i see, ok sure , i will try that |
| 15:06.37 | [TK]D-Fender | Which they should not care about |
| 15:08.49 | *** join/#asterisk serafie (~erin@nat/digium/x-rmdydznjhgqurylv) |
| 15:10.35 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:11.27 | eXcAliBuR | what happens if my digium product don't have support anymore, do I need a credit card? |
| 15:11.32 | eXcAliBuR | or do they offer free support |
| 15:11.44 | eXcAliBuR | :{ |
| 15:12.18 | file | what product? |
| 15:12.24 | eXcAliBuR | D50 |
| 15:12.25 | eXcAliBuR | phone |
| 15:12.41 | eXcAliBuR | says support not available on website under my items |
| 15:13.13 | file | what problem with it are you seeking help with? |
| 15:13.32 | eXcAliBuR | it's not provisioning with the asterisk server |
| 15:14.11 | eXcAliBuR | it works if i select the "keep old" but I can't make it use a new extension |
| 15:14.17 | eXcAliBuR | says can't fetch config |
| 15:15.19 | file | I don't know the answer, but I'll find out |
| 15:15.22 | eXcAliBuR | I'll give you all my bitcoins if you can help me |
| 15:16.02 | file | it's early in Huntsville so few are in |
| 15:16.25 | pabelanger | eXcAliBuR, all, 0.000000000000 of them? |
| 15:16.32 | Qwell | eXcAliBuR: are you sure it's able to reach your server on the port that it was configured to use? |
| 15:16.38 | eXcAliBuR | 0.00003690 |
| 15:16.40 | eXcAliBuR | ;D |
| 15:16.40 | file | or Qwell can help! |
| 15:16.47 | Qwell | file: for a consulting fee! |
| 15:17.04 | file | Qwell, you get NOTHING! |
| 15:17.04 | eXcAliBuR | well the server shows stuff happening |
| 15:17.07 | eXcAliBuR | i use port 5060 |
| 15:17.16 | Qwell | eXcAliBuR: "stuff happening" doesn't help me |
| 15:18.28 | eXcAliBuR | http://pastebin.com/k6qneeK7 |
| 15:18.41 | eXcAliBuR | thats what i see when i make the phone try to configure |
| 15:18.45 | kannan | ok , the port=15060 in peer is working ok, thanks [TK]D-Fender |
| 15:18.57 | [TK]D-Fender | kannan: You're welcome |
| 15:20.09 | eXcAliBuR | i put the res_digium_phone.conf to permission 777 thinking it might help |
| 15:20.10 | eXcAliBuR | it didn't |
| 15:20.19 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:21.29 | kannan | one more problem i have is that chanspy is geting locked on dahdi , i am not even able to request a hangup from console. If i use E flag, its not happening. but the user wants the * to jump between channels. this is asterisk 1.8.7.2 though, i need to upgrade first and see id the issue happens |
| 15:21.58 | [TK]D-Fender | kannan: Correct |
| 15:22.00 | kannan | i am not sure if user is correctly exiting with teh exit dtmf key-press too, they may just hangup |
| 15:22.03 | eXcAliBuR | i can do a reverse ssh with file or Qwell |
| 15:22.08 | eXcAliBuR | :) |
| 15:22.20 | file | DPMA is not an area I know |
| 15:23.23 | eXcAliBuR | my friend, LM didn't put much in his book on it |
| 15:23.25 | eXcAliBuR | :( |
| 15:24.19 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 15:25.18 | Qwell | eXcAliBuR: This shows the config was sent just fine. |
| 15:25.38 | eXcAliBuR | so where do we look next? |
| 15:26.04 | eXcAliBuR | i get to pick the extension from the list, then I get error fetching config |
| 15:26.17 | *** join/#asterisk yano (yano@freenode/staff/yano) |
| 15:30.34 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:30.34 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:31.58 | cusco | yellow! |
| 15:32.19 | cusco | is there any app that can playback ALL sound files in a specified dir? |
| 15:33.34 | [TK]D-Fender | ~toywy |
| 15:33.34 | infobot | rumour has it, toywy is The one you write yourself. |
| 15:34.50 | [TK]D-Fender | maybe half a dozen lines of bash for this... |
| 15:36.32 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 15:39.17 | eXcAliBuR | i also sometimes get the error, timed out contacting proxy |
| 15:39.25 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 15:41.01 | *** join/#asterisk calum_ (~calum_@cpc4-harg5-2-0-cust371.7-1.cable.virginm.net) |
| 15:48.25 | Greenlight | Or set a MOH class to point to the directory... |
| 15:48.58 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 15:50.28 | ipengineer | When upgrading to v12_b2 do we need to re-run the alembic script to change some of the column names to match the new snake_case format? |
| 15:54.28 | pabelanger | ipengineer, in fact that commit was wrong. A new alembic script should have been created, and not modified |
| 15:54.57 | pabelanger | it basically invalidated the whole concept of automated database migrations by modifying it |
| 15:55.12 | pabelanger | ipengineer, open a bug on the issue tracker |
| 15:55.16 | ipengineer | pabelanger: Ok.. I did not see that post I was just assuming we would need to run alembic and it would migrate and make any necessary changes |
| 15:55.54 | pabelanger | Yes, alembic will migrate your database, but we need to create a new changeset for it, not upgrade the existing |
| 15:56.20 | ipengineer | Ok I will file a bug with more details as I progress through the upgrade |
| 16:02.33 | *** join/#asterisk bananapie (~david@static-dslcom4-242.express.oricom.ca) |
| 16:02.50 | eXcAliBuR | the people at digium are so nice :} |
| 16:02.59 | bananapie | I accidentally launched asterisk on an SSH console, can I background it without losing the active calls ? |
| 16:03.21 | pabelanger | bananapie, should be able too |
| 16:04.01 | bananapie | any ideas how ? |
| 16:04.31 | bananapie | I think I have to do CTRL+Z which will suspend it |
| 16:04.34 | bananapie | then bg to background it |
| 16:05.17 | bananapie | nice |
| 16:05.18 | bananapie | thanks |
| 16:05.24 | pabelanger | yes |
| 16:05.58 | eXcAliBuR | do digium techs come here? |
| 16:06.23 | eXcAliBuR | this guy on the phone is prob like... oh no i got that idiot thats on IRC |
| 16:06.30 | eXcAliBuR | haha |
| 16:07.16 | bananapie | ok, I backgrounded asterisk. But I am afraid to disconnect SSH. I am not sure how to disown the process |
| 16:07.17 | [TK]D-Fender | eXcAliBuR: Not for that division really... |
| 16:07.40 | [TK]D-Fender | eXcAliBuR: You are expected to go direct... this isn't an official support channel for their phones... |
| 16:07.44 | *** join/#asterisk jpoz (~jpoz@c-24-22-121-42.hsd1.or.comcast.net) |
| 16:07.50 | [TK]D-Fender | eXcAliBuR: Though you may get lucky with the basics |
| 16:08.05 | eXcAliBuR | how do they learn their stuff? is there a book ? |
| 16:08.22 | [TK]D-Fender | eXcAliBuR: Internally... probably |
| 16:09.27 | bananapie | Nice, I closed the terminal and ssh is still running |
| 16:09.29 | *** part/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 16:15.09 | *** join/#asterisk DennisG (~DennisG@82-170-131-186.ip.telfort.nl) |
| 16:19.58 | *** join/#asterisk dxd828 (~dxd828@195.191.107.205) |
| 16:22.07 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 16:25.14 | *** join/#asterisk morcao (c1885cb9@gateway/web/freenode/ip.193.136.92.185) |
| 16:25.18 | morcao | hello guys |
| 16:25.24 | morcao | one question |
| 16:25.39 | morcao | it's possible to connect asterisk to TROPO? |
| 16:25.51 | morcao | https://www.tropo.com/ |
| 16:27.12 | pabelanger | morcao, No? Isn't the who concept of Tropo to provide you with the telephony infrastructure? |
| 16:27.19 | pabelanger | why do you want to connect asterisk? |
| 16:28.29 | mjordan | welll..... ipengineer/pabelanger: we talked about whether or not we should create a new script or update the existing. Since it was a beta release, we went with just update the existing. :-\ |
| 16:29.22 | ipengineer | mjordan: Ok.. i was just browsing jira and didn't see anything. I will leave it as is then. I just manually updated everything |
| 16:29.43 | pabelanger | mjordan, Ya, once committed to svn I think the safes way is to create another script, cannot guarantee people won't run directly from subversion. |
| 16:30.24 | mjordan | while something is in a test cycle, I'm not sure it's worth generating lots of diffs for things. Clearly once the release is made, that's not the case. |
| 16:30.50 | morcao | for connect to contact's in asterisk to telephony |
| 16:31.06 | ipengineer | mjordan: Yea. We have this problem all the time. What we do is create migrations through all of the initial stages and then when we hit a stable release we "squash" all of them into the latest version |
| 16:31.28 | [TK]D-Fender | morcao: No. |
| 16:31.44 | [TK]D-Fender | morcao: their service does not have "voice" go to you directly at all. |
| 16:31.52 | [TK]D-Fender | morcao: there is nothing to interact with on your side |
| 16:31.57 | pabelanger | Ya, but that said, if you did have automated testing of the databases, you want to test migrations. Adding code into a migration tool, then modifying said migration tool vs writing a migration seems... incorrect |
| 16:32.05 | [TK]D-Fender | morcao: Go read the book to learn what Asterisk does. |
| 16:32.07 | [TK]D-Fender | ~book |
| 16:32.07 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:32.44 | pabelanger | ipengineer, Squashing is good before a release, I'm down with that. However, it also give you a subversion history of the script _if_ people run from subversion |
| 16:32.47 | pabelanger | something to point to |
| 16:33.05 | [TK]D-Fender | morcao: and Asterisk IS the tool to connect to "telephony" |
| 16:33.47 | ipengineer | Yea.. that is the main reason we do it internally on our projects |
| 16:34.06 | pabelanger | agrees with that approach |
| 16:36.43 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 16:46.14 | weinerk | Please advise. Given: ast1.4 + conference + SIP + agi - need to be able to catch/identify DTMFs pressed. |
| 16:46.40 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 16:48.13 | *** join/#asterisk toyotapie (~david@static-dslcom4-242.express.oricom.ca) |
| 16:48.30 | toyotapie | I ran asterisk with coredump=yes, it just segfaulted. Which directory contains the core dump? |
| 16:49.18 | morcao | Yes, you is correct, but the communication between Asterisk Client and Phone Normal isn't free, yesterday you write this |
| 16:49.56 | morcao | bananapie your installation is wrong reinstall |
| 16:50.22 | morcao | and it should work |
| 16:50.33 | bananapie | what do you mean "it's wrong" ? |
| 16:51.30 | morcao | I've had this problem with the installation, try to install again |
| 16:51.52 | bananapie | core dumps or segfaulting? |
| 16:51.56 | morcao | yes |
| 16:52.14 | morcao | I think I've even had the two |
| 16:52.16 | morcao | lol |
| 16:52.35 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 16:52.49 | morcao | where to follow its installation |
| 16:52.56 | morcao | ? |
| 16:53.12 | bananapie | I tried installing it 4 different ways. From sources on centos with asterisk 10 certified, centos sources asterisk 1.8 certified, ubuntu 12.04 sources 1.8 certified and ubuntu 10.04 sources 1.8 certified and ubuntu repository. They all segfault about once every two weeks. |
| 16:53.27 | [TK]D-Fender | morcao: https://www.tropo.com/pricing/ <-- tropo isn't free either |
| 16:53.48 | [TK]D-Fender | morcao: "Tropo is 100% free during development and testing. You decide when you're ready to upgrade to production." <- how long do you think THAT will last? |
| 16:54.06 | [TK]D-Fender | morcao: Stop looking for a free lunch... |
| 16:54.59 | *** join/#asterisk anonymouz666 (~anonymouz@187.76.181.102) |
| 16:55.22 | morcao | I want to make an api that is what makes PROTO. So I'm studying the asterisk |
| 16:56.36 | morcao | ah ah I'm hungry;) |
| 16:56.37 | [TK]D-Fender | morcao: "make an API" could mean anything. that offers no scope of what it will allow, |
| 16:57.31 | *** join/#asterisk jpoz (~jpoz@ec2-184-169-152-1.us-west-1.compute.amazonaws.com) |
| 16:58.12 | morcao | yes that is my goal. I have to do something that will communicate between WebRTC and PSTN. Have any better idea? |
| 16:58.38 | morcao | worth a lunch |
| 16:58.39 | morcao | ;) |
| 16:59.40 | bananapie | when asterisk does a coredump, where is the dump file saved so I can run it in the debugger? |
| 17:00.22 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 17:00.50 | bananapie | the internet says it should be in the working directory of asterisk. |
| 17:01.58 | morcao | Any conflict with ports? |
| 17:02.17 | morcao | bananapie ? |
| 17:02.34 | *** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net) |
| 17:02.42 | bananapie | No. |
| 17:03.24 | morcao | http://linux.autostatic.com/asterisk-and-sipml5-interoperability |
| 17:03.29 | morcao | try this tutorial |
| 17:03.51 | morcao | and run the commands |
| 17:04.05 | morcao | apt-get remove --purge asterisk |
| 17:04.10 | morcao | apt-get clean |
| 17:04.15 | morcao | apt-get autoremove |
| 17:04.23 | morcao | and in the folder of asterisk |
| 17:04.28 | morcao | make uninstall-all |
| 17:04.32 | morcao | and reinstall |
| 17:08.38 | *** join/#asterisk navaismo (~navaismo@189.241.101.4) |
| 17:10.27 | morcao | [TK]D-Fender: any ideia ? |
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| 17:19.06 | anonymouz666 | morcao: how old is this article? |
| 17:19.10 | anonymouz666 | is it up to date? |
| 17:24.15 | morcao | which? |
| 17:25.03 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 17:30.30 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 17:30.52 | [TK]D-Fender | [11:58]morcaoyes that is my goal. I have to do something that will communicate between WebRTC and PSTN. Have any better idea? <- Asterisk already does this |
| 17:33.17 | morcao | Yes but not so free? right? |
| 17:33.37 | [TK]D-Fender | the PSTN side, no. |
| 17:34.35 | morcao | right, part of communicating with all already have functional |
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| 18:07.22 | drmessano | An API that will provide free PSTN connectivity? |
| 18:07.43 | Penguin | Where do I register? |
| 18:09.28 | drmessano | Must be one of those RESTful SOAP XML HTML5 things |
| 18:10.10 | Greenlight | Didn't we have one of these yesterday too, asking about *free* PSTN calls ? |
| 18:10.20 | drmessano | I'm gonna guess that since Linux is too hard, this will have to run on Windows |
| 18:10.23 | drmessano | Greenlight, same guy |
| 18:10.27 | Greenlight | Oh lol |
| 18:10.56 | drmessano | It's WebRTC <> Asterisk <> FREE <> ???? <> PSTN now |
| 18:11.13 | drmessano | Via an "API" |
| 18:11.16 | file | hmm? |
| 18:11.26 | Greenlight | AHh...so introduce more tech's and hopefully that'll somehow make the PSTN bit free. I like the logic. |
| 18:11.48 | drmessano | Greenlight, tech + glue stick = new tech |
| 18:11.58 | Greenlight | :) |
| 18:13.18 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 18:13.31 | drmessano | The biggest downside to WebRTC is that it requires something to connect it to the PSTN, and that costs money |
| 18:13.35 | drmessano | I give it 6 monhts |
| 18:13.38 | drmessano | months* |
| 18:15.14 | *** join/#asterisk karl-s (~karl.s@pool-173-51-92-61.lsanca.fios.verizon.net) |
| 18:17.40 | drmessano | It's kinda like when FXS card came out for Asterisk many years ago. For the first time, I could be a PSTN thing. It failed miserably because (1) After I signed up my neighors and interconnected, they couldn't anyone but each other, and (2) Dad kept running over the cables with the lawnmower |
| 18:18.17 | drmessano | couldn't call* |
| 18:18.25 | outtolunc | aww come on.. chromecast+4g card+tv = the future via google (haha) |
| 18:18.25 | mjordan | drmessano: well, they did solve problem (2) |
| 18:18.52 | navaismo | 6 months to webrtc only? uh i was hoping a couple of years |
| 18:19.24 | file | it's amusing... |
| 18:19.44 | file | the die-hard WebRTC folk said 6 months in the beginning |
| 18:21.16 | navaismo | outtolunc, i want a chromecast but now im afraid becasue lG logg everything and send to his server, i guess al vendors now do that :( |
| 18:21.50 | navaismo | dont want the file my_super_pr0nvid.mp4 go public |
| 18:22.28 | outtolunc | for $35 its worth it just to open it up and play with it |
| 18:22.39 | outtolunc | kape already got one.. |
| 18:23.06 | outtolunc | preview sdk supposedly available |
| 18:23.12 | drmessano | file, I should be able to talk to anyone in the world, via PSTN/Skype/SIP/H323/Carrier Pigeon via WebRTC because it runs in my free browser. Otherwise, I will go back to AOL Instant Messenger |
| 18:23.52 | navaismo | ICQ |
| 18:24.10 | drmessano | 39882381 <-- add me |
| 18:25.00 | drmessano | Or hack it. I haven't signed into it for years. The password may still be drowssap |
| 18:25.02 | outtolunc | wow, big num |
| 18:25.33 | navaismo | lol |
| 18:25.34 | drmessano | That's a 1997 ICQ number |
| 18:25.39 | drmessano | Before the purge |
| 18:25.48 | mjordan | I don't have mine any more. Would have been around that time frame. |
| 18:26.13 | drmessano | I hated that fscking flower |
| 18:26.17 | mjordan | heh |
| 18:26.18 | outtolunc | i was under 200k and that was within months of the startup |
| 18:26.31 | drmessano | Sitting on dialup, hoping it would stop spinning |
| 18:26.36 | mjordan | changing colors |
| 18:26.36 | drmessano | COME ON, CONNECT.. PLEASE |
| 18:26.42 | navaismo | i was too youg to use it hahahaha |
| 18:26.43 | mjordan | please don't go red, please don't go red, please don't go red |
| 18:26.52 | drmessano | HAHA yep |
| 18:27.29 | drmessano | It was like "Press Your Luck" |
| 18:27.32 | outtolunc | i love it when i go to a gas station and pay inside.. i hear 'ut oh' |
| 18:27.36 | drmessano | NO WHAMMY NO WHAMMY NO WHAMMY |
| 18:27.39 | drmessano | AHHHH |
| 18:27.55 | drmessano | outtolunc, lol |
| 18:27.57 | navaismo | digging to find the old external modem |
| 18:28.03 | drmessano | I forgot all about the ut oh |
| 18:28.46 | drmessano | "Your version of ICQ is out of date. You haven't updated in 45 minutes since the last upgrade" |
| 18:29.21 | paulc | Haha.. ICQ.. those were the days.. |
| 18:29.35 | paulc | I had a 7 digit number too.. oooh prestige (but not really) |
| 18:30.06 | outtolunc | haha.. still works |
| 18:30.52 | navaismo | found it: Encore ENF656 data-fax-voice modem with 9 leds and a big serial port. Wondering if works |
| 18:31.24 | outtolunc | funny.. all my ~friends~ are offline ;) |
| 18:31.35 | paulc | Can you sign in to ICQ on the web somewhere? |
| 18:32.39 | drmessano | outtolunc, lol |
| 18:32.45 | drmessano | They have mobile apps |
| 18:32.47 | drmessano | Like MySpace does |
| 18:33.12 | drmessano | lol |
| 18:33.36 | drmessano | I had ICQ for IOS a few years ago. Signed in, giggled like a 9 yr old, signed out |
| 18:34.51 | outtolunc | you had to bring up giggling eh |
| 18:36.47 | drmessano | The ICQ iOS app was last updated in April 2013. Guess they're falling behind |
| 18:37.10 | drmessano | They need to work a little harder if they're going to stay relevant |
| 18:38.10 | drmessano | OHHH and ICQ offers FREE VOICE CALLS now |
| 18:38.34 | *** join/#asterisk jmmills (jason@204.16.200.135) |
| 18:39.23 | drmessano | submits new ticker in Asterisk JIRA - Feature Request - ICQ module so I can call my ICQ buddies. THX |
| 18:39.30 | drmessano | ticket* |
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| 18:40.12 | navaismo | can i use taht with webrtc or my deskphone? |
| 18:40.52 | drmessano | navaismo, WebRTC <> Phone <> Firefox <> Asterisk <> ICQ <--- YES |
| 18:41.01 | drmessano | Seems legit |
| 18:41.59 | navaismo | and a dongle? can i install it and use to call mobiles from my cell phone? |
| 18:42.36 | eirirs | ICQ ? what ? It's still in use? |
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| 18:47.46 | drmessano | navaismo, sure |
| 18:48.31 | drmessano | eirirs, ICQ is still around kinda like how Trixbox is still around. You can download and use it, but why... |
| 18:50.11 | eirirs | hehe |
| 18:50.15 | mjordan | huh. |
| 18:50.17 | eirirs | "because I can..." |
| 18:50.23 | mjordan | Out of intellectual curiousity, how is ICQ making voice calls |
| 18:50.43 | navaismo | awesome im going to use my 2 Rpi to provide telephony services! |
| 18:50.57 | [TK]D-Fender | mjordan: natively, not ... TTS+AVR :) |
| 18:51.19 | drmessano | Does OSCAR support voice? |
| 18:51.51 | [TK]D-Fender | drmessano: Don't forget to use Dr. Sbaitso for the TTS ;) |
| 18:51.57 | drmessano | lol |
| 18:52.13 | drmessano | Ok, I JFGI |
| 18:53.19 | drmessano | AIM used/uses OSCAR + SIP/RTP for voice chat. |
| 18:53.26 | drmessano | Maybe ICQ did the same |
| 18:54.04 | drmessano | I dont know how much technology transferred to the new owners when AOL sold ICQ, but I seem to remember it having "voice chat" |
| 18:54.57 | drmessano | Maybe they had to wait 6 or 7 years for the ICQ SIP/RTP server to arrive in Siberia from AOL's HQ and they're now turning it on?? |
| 18:55.40 | mjordan | I wonder if everyone building WebRTC clients that have text + voice + video realize they're implementing ICQ in a browser (which also exists) |
| 18:55.51 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 18:56.14 | navaismo | :O |
| 18:56.42 | drmessano | mjordan, boom! |
| 18:58.05 | drmessano | Ok, it hasnt been THAT long since ICQ was sold to the Russian mob.. 2010 |
| 18:58.57 | drmessano | $187.5 millions dollars. I guess it's still big in Brazil |
| 18:59.03 | drmessano | and Russia |
| 18:59.12 | mjordan | I'd do a lot for 187 million. Just sayin'. |
| 18:59.26 | Penguin | prepares a request |
| 19:00.38 | drmessano | Hey now |
| 19:01.42 | navaismo | i can give a kidney for that amount of money without problem |
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| 19:02.36 | drmessano | I wouldn't quit my job if I won that kind of money |
| 19:02.48 | drmessano | Just the leverage... "I could QUIT, you know?" |
| 19:02.57 | drmessano | That would be great |
| 19:03.15 | outtolunc | roll in your own server rack.. ;) |
| 19:05.07 | drmessano | Nah, our connectivity sucks here :) |
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| 19:16.41 | anonymouz666 | drmessano: ICQ? Brazil? |
| 19:16.51 | *** join/#asterisk DennisG (~DennisG@82-170-131-186.ip.telfort.nl) |
| 19:17.34 | anonymouz666 | it was big... until MSN landed in 2002. |
| 19:17.47 | drmessano | anonymouz666, I had to pick some large South American market. Brazil came to mind. Isn't Orkut still huge there? |
| 19:18.21 | anonymouz666 | Orkut was a big resistence |
| 19:19.44 | anonymouz666 | but... not anymore. |
| 19:20.07 | *** join/#asterisk SushiB (~Thunderbi@189.215.47.77.cable.dyn.cableonline.com.mx) |
| 19:20.13 | anonymouz666 | facebook owned |
| 19:20.39 | drmessano | I thought Orkut was awesome because you signed up for MySpace, and it was all about Tom... You signed up for Orkut and there was this guy named Orkut who reminded me of one of the characters from Night at the Roxbury and we all wanted to be his buddy |
| 19:21.36 | drmessano | Currently, Orkut Büyükkökten is a product manager at Google. He also is a certified masseur, an avid ballroom dancer and likes to make chocolate fondue <--- From Wikipedia. How could you not LOVE that guy? |
| 19:21.56 | anonymouz666 | hehe |
| 19:22.43 | [TK]D-Fender | Because I'm hetero-sexual? :) |
| 19:22.46 | drmessano | If you Google Image search from him, there's a bunch of pics of him just being a freakin cool dude |
| 19:22.55 | anonymouz666 | whatsapp here is becoming HUGE |
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| 19:26.00 | drmessano | Really? |
| 19:26.23 | drmessano | Everyone I know hates it. I have tried to get people to use it, but bleh |
| 19:26.42 | drmessano | It could just be the part about adding me |
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| 19:49.58 | HumpyDumpy | It's ok everyone, humpy is here now |
| 19:50.06 | joako | Are bug reports against Asterisk 1.6.x still worked on? |
| 19:50.15 | Penguin | No. |
| 19:50.25 | [TK]D-Fender | not since a long time |
| 19:50.32 | Penguin | 1.6.x branches are all past EOL. |
| 19:51.30 | joako | Is there a newer version that can use unmodified 1.6 configurations? |
| 19:51.58 | [TK]D-Fender | joako: Things change. |
| 19:52.22 | [TK]D-Fender | joako: so in general... "no" |
| 19:52.38 | [TK]D-Fender | joako: With any luck you will need to change SEOMTHING. |
| 19:52.41 | [TK]D-Fender | something* |
| 19:52.48 | [TK]D-Fender | joako: Just a question of how many |
| 19:53.03 | Penguin | Mostly dial plan applications, I'd imagine. |
| 19:54.10 | [TK]D-Fender | mostly |
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| 19:54.44 | Penguin | A few items in sip.conf are different, if you only go to 1.8. Probably more if you go to 10+. |
| 19:55.11 | joako | Penguin, That´s mostly what concerns me. I guess I´ll keep 1.6 for now until I find a larger bug. Current issue is Asterisk segfaults if there are too many voicemails in IMAP |
| 19:55.21 | [TK]D-Fender | If you're going to change... don't waste time on the stop-gap measure... go right to 11. |
| 19:57.09 | Penguin | I'll probably upgrade to 11 by the time 13 is released. |
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| 20:02.52 | joako | I just placed a call with a 16 year old mobile phone... I think Asterisk 1.6 can run fine for quite a while. |
| 20:03.02 | HumpyDumpy | looks around for someone to bite |
| 20:03.13 | HumpyDumpy | eyes defender |
| 20:04.01 | [TK]D-Fender | hands HumpyDumpy a spoon |
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| 20:33.12 | redotis | asterisk -rx "sip show channelstats" Shows natted ips and public addresses? Is this normal? |
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| 21:14.31 | drmessano | What is this Asterisk 1.6 you speak of? |
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| 22:09.05 | redotis | asterisk -rx "sip show channelstats" Shows natted ips and public addresses? Is this normal? |
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| 22:42.21 | WIMPy | Is (lib)amrnb required and not optional? Or am I missing something? |
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| 23:44.33 | arctanx | Hi all. I'm using a snom phone to page a bunch of loudspeakers. If I use A(tt-weasels) I get a nice announce sound and can keep speaking afterward. If I use any GSM file I created myself using sox it plays fine but the snom microphone no longer gets connected to the speakers so I only get silence afterwards |
| 23:45.28 | arctanx | Could anyone please give me any insights into where the problem might be? It smells like a file format thing but it's identical according to soxi |
| 23:46.20 | paulc | Have you tried with a G.711 format audio file? |
| 23:46.33 | paulc | (because that's probably what your phones are using?) |
| 23:46.54 | paulc | that said.. I'd imagine the snom uses it be default so Asterisk would transcode the gsm file to G.711 |
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| 23:49.46 | arctanx | the snom is offering G.711 as its codec 1 apparently |
| 23:49.52 | arctanx | u, then a |
| 23:50.31 | arctanx | tries again with ulaw |
| 23:54.55 | arctanx | Yeah it doesn't like that |
| 23:55.04 | arctanx | Just clicky noise |
| 23:56.39 | paulc | Hmm.. that's weird.. I know you can sometimes get a click at the start of ulaw if you save it as ulaw-with-header as opposed to headerless-ulaw.. |
| 23:56.41 | navaismo | hmm where is the rfc2833 mode in pjsip? Is not a valid mode anymore? |
| 23:56.47 | paulc | (it's like the WAV metadata) |
| 23:58.28 | arctanx | interesting |
| 23:59.01 | arctanx | That might be related to the clicky business. I've also tried converting my file to a compatible .wav. It also plays fine but still doesn't let me talk after |