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| 00:45.00 | fabiobik | hello guys |
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| 00:45.21 | fabiobik | does anyone here have played with 3g dongles? |
| 00:45.47 | fabiobik | i think ive configurated all right |
| 00:46.18 | fabiobik | now im trying to call the number and play hello worl |
| 00:46.20 | fabiobik | world |
| 00:47.04 | fabiobik | exten => s,1,Answer |
| 00:47.29 | fabiobik | exten => s,n,Playback(hello-world) |
| 00:47.55 | fabiobik | i call to the number but nothing happen |
| 00:48.49 | ChannelZ | So you actually see anything on the console when you attempt to call in? |
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| 00:50.43 | fabiobik | ChanServ, nop |
| 00:50.52 | fabiobik | ChannelZ |
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| 00:58.11 | fabiobik | ChannelZ: nop |
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| 01:05.19 | ChannelZ | Guess your hardware is not quite configured/interfacing right but I don't know anything of it |
| 01:09.52 | [TK]D-Fender | fabiobik: We have no idea if you've configured * right to process whatever calls may even be making it there. |
| 01:10.33 | [TK]D-Fender | fabiobik: You haven't told us what you're using to call * with. 3G is a set of wireless data bands so I'd presume you mean some sort of VoIP client. |
| 01:10.47 | fabiobik | sorry |
| 01:10.55 | fabiobik | its a Huawei E160X |
| 01:10.57 | [TK]D-Fender | fabiobik: But we'd need to knoow specifically so we can actually debug the call |
| 01:11.19 | [TK]D-Fender | fabiobik: I don't need to know the 3G dongle. that is just a DATA source |
| 01:11.30 | [TK]D-Fender | fabiobik: You are presumably using it for VOIP traffic. |
| 01:11.38 | [TK]D-Fender | I asked what is calling * exactly |
| 01:12.00 | lorsungcu | a phone, obviouslty. |
| 01:12.01 | fabiobik | nothing i want to play a menu in future |
| 01:12.28 | fabiobik | im calling from a phone to a dongle of corse |
| 01:12.36 | lorsungcu | oh of course |
| 01:12.57 | [TK]D-Fender | fabiobik:o you are looking to use the 3G as a VOICE device? |
| 01:13.04 | fabiobik | yes |
| 01:13.20 | [TK]D-Fender | fabiobik: Well show us what you've done to set it up and so that * knows about it |
| 01:13.24 | [TK]D-Fender | ~pb |
| 01:13.24 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 01:13.24 | fabiobik | ive seen and the voice its enabled |
| 01:13.26 | [TK]D-Fender | ^^^^ |
| 01:13.55 | fabiobik | what you want to paste friend? |
| 01:14.19 | fabiobik | *to i |
| 01:14.34 | fabiobik | extensions.conf and dongle.conf? |
| 01:14.49 | [TK]D-Fender | fabiobik: SAhow us it's configuration, and some status dump to see that it's active |
| 01:15.20 | fabiobik | can you tell me what you want in specific? in noob at asterisk |
| 01:15.39 | fabiobik | tell me what command i should paste |
| 01:16.05 | [TK]D-Fender | fabiobik: I asked you show us what YOU did. YOU must have set up config files so that * even knows about it. Show us those configs |
| 01:16.18 | fabiobik | sure |
| 01:16.19 | fabiobik | 1 sec |
| 01:23.57 | fabiobik | ... |
| 01:24.12 | fabiobik | im instaling that thing nautilus-pastebin |
| 01:24.37 | fabiobik | but aptitude have started maintain too -_- |
| 01:26.29 | fabiobik | So here is my dongle.conf |
| 01:26.31 | fabiobik | http://www.voip-info.org/wiki/view/Asterisk+CLI |
| 01:26.34 | fabiobik | fuck |
| 01:26.41 | fabiobik | http://www.voip-info.org/wiki/view/Asterisk+CLI |
| 01:26.45 | fabiobik | ... |
| 01:27.14 | fabiobik | paste.debian.net/244097/ |
| 01:28.10 | fabiobik | paste.debian.net/244098/ |
| 01:28.23 | fabiobik | last one is the extensions.conf |
| 01:30.14 | [TK]D-Fender | What is the first one exactly? |
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| 01:32.14 | fabiobik | the first one is a dongle configuration |
| 01:32.20 | [TK]D-Fender | what file exactly? |
| 01:32.29 | fabiobik | dongle.conf |
| 01:32.43 | [TK]D-Fender | that is not the right name for it from what I've seen |
| 01:32.58 | [TK]D-Fender | Actually.. |
| 01:33.04 | [TK]D-Fender | see another reference that it could be... |
| 01:33.07 | [TK]D-Fender | hmm |
| 01:33.11 | [TK]D-Fender | ok, on to the contents |
| 01:33.27 | fabiobik | : ) |
| 01:34.04 | [TK]D-Fender | fabiobik: Both the context AND the exten in your dongle.conf fail to match the DIALPLAN in your extensions.conf |
| 01:34.15 | [TK]D-Fender | exten=+351913496442 |
| 01:34.22 | [TK]D-Fender | context=default |
| 01:34.29 | [TK]D-Fender | This is not what you have in extensions.conf |
| 01:34.37 | [TK]D-Fender | So that should automatically fail |
| 01:34.53 | [TK]D-Fender | audio=/dev/ttyUSB2; tty port for audio connection; no default value |
| 01:34.55 | [TK]D-Fender | data=/dev/ttyUSB3; tty port for AT commands; no default value |
| 01:35.08 | [TK]D-Fender | Also verify if these are indeed the proper device names |
| 01:36.15 | fabiobik | where have you seen context=default |
| 01:37.05 | [TK]D-Fender | fabiobik: http://paste.debian.net/244097/ <----- |
| 01:37.16 | [TK]D-Fender | line #42 |
| 01:40.35 | jeev | fender, remember that issue i had within like 24 seconds or something, an incoming call would drop? |
| 01:40.43 | [TK]D-Fender | not really |
| 01:41.33 | jeev | ah, incoming calls would go past the ring group, like the third attempt, anyway. i'm not sure if i told you but the toll carrier was doing it to prevent possible hung calls. |
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| 01:41.57 | [TK]D-Fender | :/ |
| 01:42.05 | jeev | dude but how do you guys deal with some clients |
| 01:42.23 | jeev | one sent me an email claiming there was a glitch in the system because someone was calling a random out of the lata did and it was coming to them.. |
| 01:42.34 | jeev | i verified with the telco that it was just coming in.. nothing i could do, someone was probably forwarding it. |
| 01:46.46 | [TK]D-Fender | Can't live with 'em .... can't shoot 'em |
| 01:47.18 | jeev | i guess if i were in a red state i could pull a stand your ground in their office/ |
| 01:47.21 | jeev | ? |
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| 01:52.04 | TJNII | I have a server that has begun acting up. Server is behind NAT, connecting to a ITSP outside. RTP and SIP ports are NATted back to the * server. "nat=yes, externip, and localnet" are configured in sip.conf. When a call is attempted, SIP handshaking completes and then * begins sending RTP packets to externip, not the IP of the ITSP. Any wise words? |
| 01:52.53 | [TK]D-Fender | TJNII: that's the START of what you need to set. Then there are all of your peer settings |
| 01:55.48 | TJNII | Peer settings are set, this is a formerly working box. Can you enlighten me on which setting would cause * to send RTP to its ip (externip) rather than the peer? |
| 01:56.09 | fabiobik | what is the max verbose of asterisk? |
| 01:56.11 | fabiobik | 6? |
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| 01:57.28 | [TK]D-Fender | fabiobik: Pracically speaking? yes I think. I always use 10 personally... |
| 01:57.48 | [TK]D-Fender | TJNII: Show us your configs and the call with SIP debug enabled |
| 01:57.50 | [TK]D-Fender | ~pb |
| 01:57.50 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 01:57.52 | [TK]D-Fender | ^^^ |
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| 02:55.49 | leifmadsen | fabiobik: max verbose, useful, is probably around 10 or so. You can set it to 999 or whatever though. |
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| 04:53.40 | fabiobik | what i need? I want to make something like this gsm gateway -> asterisk -> sip -> destination number |
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| 04:54.11 | fabiobik | when someone calls to asterisk we ear something like enter the dial number |
| 04:54.24 | fabiobik | how to accomplish something like that? |
| 04:54.34 | [TK]D-Fender | fabiobik: I already told you 2 things you did wrong in your config and I haven't seen you come back with them fixed |
| 04:54.53 | fabiobik | fender the problem is solved |
| 04:54.57 | [TK]D-Fender | fabiobik: And if you want to play a sound to your caller... then go play a sound |
| 04:55.01 | fabiobik | now i want to try this approach |
| 04:55.06 | [TK]D-Fender | fabiobik: "core show application playback" <- |
| 04:55.10 | [TK]D-Fender | ~book |
| 04:55.10 | 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 |
| 04:55.12 | [TK]D-Fender | ^^ |
| 04:55.24 | fabiobik | i already know that |
| 04:56.03 | [TK]D-Fender | then make a basic IVR or use "read" |
| 04:56.27 | fabiobik | fender yes that i know |
| 04:56.33 | fabiobik | and then? |
| 04:56.41 | [TK]D-Fender | And then go process the result |
| 04:56.51 | [TK]D-Fender | IVR's use WaitExten() |
| 04:57.03 | [TK]D-Fender | and jump to the exten they enter. |
| 04:57.09 | [TK]D-Fender | You need to read your dialplan basics |
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| 10:05.01 | dipanjan | Is it possible to use multiple USB dongles simultaneously for calls with chan_dongle? |
| 10:09.17 | WIMPy | I've seen that in the configuration. |
| 10:15.05 | dipanjan | WIMPy: you mean dongle.conf? |
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| 15:17.40 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 15:26.07 | Helmut_ | hi |
| 15:28.15 | WIMPy | lo |
| 15:28.43 | Helmut_ | I just started playin around with a asterisk vm and have some questions |
| 15:30.17 | Helmut_ | I did the introduction and the server is working. But when I want to make a call after I press "Answer" the call is cancelled |
| 15:30.44 | Helmut_ | do u know where I have to work on? |
| 15:31.13 | Helmut_ | internal and external calls |
| 15:31.41 | WIMPy | Look at the console. Turn up verbose to at least 3. |
| 15:32.55 | Helmut_ | [Mar 24 16:32:21] WARNING[1355][C-00000071]: chan_sip.c:22807 handle_response_invite: just did sched_add waitid(2695) for sip_reinvite_retry for dialog 3e7f1102044200a661c4d158475cd017@10.5.1.107:5060 in handle_response_invite -- Got SIP response 486 "Busy here" back from 10.5.1.105:33749 == Spawn extension (meine-telefone, 2001, 1) exited non-zero on 'SIP/2000-0000004a' |
| 15:33.32 | WIMPy | "Busy here" does not look like an answer. |
| 15:34.08 | Helmut_ | it appears when I press Answer on the ringing phone |
| 15:34.34 | WIMPy | That "Busy here" is what the phone said. |
| 15:34.43 | Helmut_ | y |
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| 15:36.16 | Helmut_ | when I press cancell the mailbox is working |
| 15:36.53 | WIMPy | You could try to disable reinvites. Looks like that phone isn;t handling them propperly. |
| 15:37.31 | Helmut_ | its on all phones. even external numbers |
| 15:37.56 | WIMPy | With the same warning? |
| 15:38.02 | Helmut_ | sec |
| 15:38.53 | WIMPy | And what does external number mean here? |
| 15:40.44 | Helmut_ | outside the LAN |
| 15:40.56 | Helmut_ | a normal telephone number |
| 15:41.22 | WIMPy | To or from? and what's the other end? |
| 15:41.58 | Helmut_ | <PROTECTED> |
| 15:41.58 | WIMPy | And how are you connected to the PSTN? |
| 15:42.32 | WIMPy | That looks like they don't like you. |
| 15:44.40 | Helmut_ | Finally I only want to create a call-forwarding. Incoming on a different sip-provider than outgoing. |
| 15:45.47 | Helmut_ | Until now it rings only like it should |
| 15:46.20 | WIMPy | You should test them one by one. |
| 15:47.30 | Helmut_ | First I only tested internal calls without an external provider. the problem is already here |
| 15:51.37 | Helmut_ | may there be a port issue? |
| 16:00.10 | WIMPy | That does not look like one. |
| 16:00.41 | WIMPy | You can turn on sip debug and take a closer look at what's going on. |
| 16:05.58 | Helmut_ | how do I do it? |
| 16:06.19 | WIMPy | 'sip set debug on' |
| 16:07.58 | Helmut_ | <--- Transmitting (no NAT) to 10.5.1.105:36367 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.5.1.105:36367;branch=z9hG4bK5f03d8eadffb54ecbeb45d578ef22662343036;rport;received=10.5.1.105 From: "2000" <sip:2000@10.5.1.107>;tag=650741032 To: "2000" <sip:2000@10.5.1.107>;tag=as2bedcd22 |
| 16:08.51 | WIMPy | "Not Found" means the called number doesn;t exist. |
| 16:09.32 | WIMPy | But you should have seen that without sip debug. |
| 16:09.41 | [TK]D-Fender | We don't see what that is a response TO |
| 16:09.47 | [TK]D-Fender | PASTEBIN the complete debug |
| 16:09.48 | [TK]D-Fender | ~pb |
| 16:09.49 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 16:09.51 | [TK]D-Fender | ^^^ |
| 16:14.17 | Helmut_ | http://pastebin.com/NFtpQvdC |
| 16:15.22 | WIMPy | And now again with a failed call. |
| 16:15.27 | [TK]D-Fender | As suspected that has nothing to do with a CALL |
| 16:22.56 | Helmut_ | ok I hope this provides better info http://pastebin.com/8gdZBcqd |
| 16:25.08 | [TK]D-Fender | I see a reinvite failing, and not quite a complete call |
| 16:25.19 | [TK]D-Fender | get it from the start |
| 16:26.36 | Helmut_ | what does that mean? |
| 16:26.55 | WIMPy | The beginning was missing. |
| 16:27.19 | WIMPy | Start sip debug and then copy everyhting from before you start the call until after it's finished. |
| 16:27.38 | WIMPy | The beginning is missing. |
| 16:29.55 | Helmut_ | okay, this time I tried to be fast. Maybe the important things didnt vanish http://pastebin.com/ncyNarSR |
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| 16:30.57 | WIMPy | Log to a file or use tee or whatever. |
| 16:31.26 | WIMPy | SIP debugs are quite long. |
| 16:31.42 | [TK]D-Fender | Helmut_: get a bigger scroll-back buffer |
| 16:31.51 | [TK]D-Fender | Helmut_: and that is not the start of the call |
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| 16:45.58 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 16:46.10 | Helmut_ | sry, I can't find an appropriate way to increase the scroll-back puffer so far |
| 16:47.05 | *** join/#asterisk aynam (29cfb816@gateway/web/freenode/ip.41.207.184.22) |
| 16:49.02 | igcewieling | check the asterisk log files, the sip debug should be logged there as well. |
| 16:49.09 | WIMPy | use tee |
| 16:49.22 | igcewieling | check /etc/asteisk/logger.conf make sure debug is enabled for the log file. |
| 16:49.55 | WIMPy | or use a better terminal. |
| 16:50.05 | aynam | hello i have just installed asterisk and freepbx and i just get a blank page on .../admin/config.php page. what to do to correct that |
| 16:50.28 | WIMPy | aynam: Ask in #freepbx |
| 16:50.50 | aynam | WIMPy: ok |
| 16:59.34 | *** join/#asterisk resist0r (resist0r@unaffiliated/mrresist0r) |
| 17:02.30 | Helmut_ | Now I got it. Full logfile from beginning. http://pastebin.com/tsgECwJZ thx for patience |
| 17:03.25 | *** join/#asterisk vedic (~V@183.82.86.30) |
| 17:05.19 | vedic | How to say a number something like: 101.567 ? There is a saynumber command for IVR, but how about decimal numbers? |
| 17:05.52 | WIMPy | You have to split that yourself. |
| 17:06.21 | WIMPy | Or actually it might be possible to tweak the configuration? |
| 17:07.03 | vedic | WIMPy: I only have one idea that split it a decimal and then use say number and say digit with 'dot' in between. So total 3 prompts |
| 17:07.13 | vedic | WIMPy: any other trick? |
| 17:08.40 | WIMPy | Helmut_: You seem to have a networking issue between your Asterisk and sip/2000. |
| 17:08.57 | WIMPy | vedic: Nope |
| 17:09.28 | WIMPy | Helmut_: It's just retransmitting and not getting any answer. |
| 17:10.04 | WIMPy | Or not until after the fifth try. |
| 17:13.11 | Helmut_ | an idea where to work on? |
| 17:14.09 | WIMPy | Find out why it takes so long for those applications to communicate. |
| 17:14.34 | *** join/#asterisk hehol (~Adium@ip-78-94-14-150.unitymediagroup.de) |
| 17:15.35 | [TK]D-Fender | Helmut_: Retransmitting #3 (no NAT) to 10.5.1.105:36367: |
| 17:15.37 | [TK]D-Fender | INVITE sip:2000@10.5.1.105:36367;transport=udp SIP/2.0 |
| 17:15.48 | [TK]D-Fender | Helmut_: SIP/2000 is not responding at all |
| 17:16.17 | *** join/#asterisk raden (~Jon@24-240-51-238.dhcp.stpt.wi.charter.com) |
| 17:16.21 | [TK]D-Fender | or massively slowly |
| 17:16.29 | raden | [Mar 24 12:14:18] ERROR[5240]: res_config_sqlite.c:842 cdr_handler: database is full |
| 17:16.31 | raden | ? |
| 17:19.30 | Helmut_ | the test clients are connected via wlan |
| 17:19.54 | [TK]D-Fender | Helmut_: Check firewalls on the clients as well |
| 17:20.12 | [TK]D-Fender | but that got multiple re-transmits. I'm betting on high PL |
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| 17:31.00 | Helmut_ | the phones have no firewall what is high PL? |
| 17:31.27 | [TK]D-Fender | Packet Loss |
| 17:31.58 | [TK]D-Fender | on your WLAN |
| 17:32.30 | Helmut_ | I tested in front of the access point without changes |
| 17:34.07 | Helmut_ | the phones work configured directly to a sip-provider |
| 17:35.21 | [TK]D-Fender | Helmut_: set "directmedia=no" on your peers and retest. They are very ppor at responding and I was watching the reinvite attempt bounce around as well. |
| 17:35.35 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
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| 17:37.50 | Helmut_ | in sip.conf? |
| 17:38.21 | WIMPy | With that kind of start, I'd expect prettu bad call quality. |
| 17:38.47 | WIMPy | s/u/y/ |
| 17:39.45 | [TK]D-Fender | Helmut_: In your peer entries there, yes |
| 17:46.57 | Helmut_ | Its WORKING XD |
| 17:47.05 | Helmut_ | Thank You! |
| 17:50.44 | vedic | When I update my linux which also updates kernel, it breaks asterisk. How to avoid breaking asterisk but still allow kernel updates |
| 17:51.06 | ChannelZ | are you using DAHDI? |
| 17:51.07 | [TK]D-Fender | Helmut_: could be they were panicking at their own attempts to talk direct. If your PL is that bad from device -> * then from device->device it would be double |
| 17:51.23 | [TK]D-Fender | Helmut_: Your phones / environment sound extremely flakey... |
| 17:54.11 | Helmut_ | I tested on two Android phones and a VM. Finally I only want to get a call-forwarding working and the phones don't have to stay in the environment |
| 17:56.52 | [TK]D-Fender | Alrighty... |
| 18:02.02 | vedic | ChannelZ: yea |
| 18:02.32 | vedic | ChannelZ: with wanpipe drivers |
| 18:02.44 | ChannelZ | vedic: that's why, DAHDI are kernel drivers. Update the kernel, you need drivers that match. |
| 18:03.25 | vedic | ChannelZ: so if I update the kernel, I would need to compile dahdi and wanpipe again? |
| 18:03.52 | WIMPy | yes |
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| 18:27.40 | lvlinux | anybody familiar in particular w the Polycom IP430? |
| 18:28.05 | dr0ck | my grampa had one of those |
| 18:29.20 | lvlinux | lol yes I know it's not new stuff |
| 18:29.45 | lvlinux | was mainly just wondering about the wideband on it---it's not labeled "HD Voice" but does support g.722 |
| 18:30.27 | lvlinux | does that mean that the analog parts of the phone aren't optimized for full band audio, or just that they hadn't started marketing it as HD Voice yet? |
| 18:30.28 | dr0ck | gotta buy HD handset |
| 18:30.55 | dr0ck | i think they are 30ish |
| 18:31.01 | lvlinux | ahh, so it's like the Snom 300 series, where the phone supports it but the handset still cuts the frequencies. Ok i see. |
| 18:31.34 | lvlinux | what about the speakerphone? Would it be the same as the handset? you obviously can't upgrade that |
| 18:33.27 | [TK]D-Fender | lvlinux: I see no G.722 support for it |
| 18:33.35 | WIMPy | Are you sure the HD handsets aren't some kind of snake oil? |
| 18:33.41 | [TK]D-Fender | lvlinux: Not on their data sheets anywhere |
| 18:34.17 | [TK]D-Fender | the first HD was the IP 650 |
| 18:34.41 | [TK]D-Fender | 335/450/550/560/650/670 |
| 18:34.48 | lvlinux | yes i know it is not publicly documented but several people have confirmed that g.722 can be enabled on the phone via the config. |
| 18:35.20 | [TK]D-Fender | and you see it negotiated successfully? |
| 18:35.43 | lvlinux | I don't have the phone yet-- just won a cheap auction for 3 of them though. |
| 18:35.45 | [TK]D-Fender | lvlinux: Perhaps newer firmware enabled it once they found the DSP had enough juice for it. |
| 18:35.52 | lvlinux | maybe |
| 18:36.02 | [TK]D-Fender | lvlinux: I'd get real proof before running on that expectation |
| 18:36.19 | [TK]D-Fender | lvlinux: They didn't have any G.722 before teh 650 |
| 18:36.38 | lvlinux | well i didn't buy them exclusively for that feature, but at $50 shipped for 3 of them I figured I couldn't go wrong. |
| 18:36.48 | [TK]D-Fender | They are solid phones |
| 18:36.52 | [TK]D-Fender | I've got a bunch |
| 18:36.52 | lvlinux | at worst I've got 3 decent phones for that price :-D |
| 18:37.03 | [TK]D-Fender | $50 each is still a good price |
| 18:37.21 | lvlinux | no not each I got 3 for $50 |
| 18:37.27 | lvlinux | :-D |
| 18:37.36 | [TK]D-Fender | I saw that |
| 18:37.42 | [TK]D-Fender | just makes your deal insame |
| 18:37.47 | [TK]D-Fender | insane* |
| 18:38.15 | lvlinux | ya I was surprised I got them for that little. |
| 18:38.31 | [TK]D-Fender | Don't worry.. surprises come in boxes too ;) |
| 18:38.45 | lvlinux | hehe |
| 18:39.05 | lvlinux | I'm going to replace my old 7960 with one of them probably. |
| 18:52.07 | vedic | Is there default prompt for 'dot' or decimal point? |
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| 18:52.36 | vedic | As I can use default voice for digits, I want to use the default voice for 'dot' or decimal point as well |
| 18:53.40 | WIMPy | It's there as a "letter". |
| 18:56.20 | vedic | WIMPy: thank yo |
| 18:56.22 | vedic | you |
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| 19:11.40 | Helmut_ | btw my call forwarding is working fine now! job done, thank you for support and have a nice evening. |
| 19:12.37 | [TK]D-Fender | You're welcome |
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| 19:36.33 | drmessano | Anyone using BRIA for iOS and made video actually work? |
| 19:39.38 | file | I heard video is a fad, like the internet |
| 19:56.11 | igcewieling | file: we can only hope they are right. |
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| 20:07.42 | smkelly | file: Moo. |
| 20:07.54 | file | smkelly, WHAT WHAT WHATTTTTTT |
| 20:12.57 | smkelly | So this file guy once told me that AEL is not supported/maintained. Can I trust this guy? Would it be crazy to make a new dialplan in AEL? |
| 20:14.04 | smkelly | I mean, some of the examples in the Asterisk wiki even use AEL |
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| 20:25.09 | drmessano | file, video is just audio with pictures |
| 20:26.43 | drmessano | That's the standard radio engineering response when someone asks if you've worked in TV. "How hard can it be? It's just Radio with pictures" |
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