| 00:00.51 | floren | the phone log while placing a call: http://codepad.org/5il8dB9s |
| 00:02.11 | floren | anyways, i think we had enough for tonight guys :) |
| 00:02.35 | floren | i really thank you for taking the time to help out, much apreciated |
| 00:03.33 | ChannelZ | well it claims to be sending something to 191.168.1.9 to port 5060 UDP, but if you have 'sip set debug on' and see *nothing* then it's some other network issue. |
| 00:04.01 | floren | i'm going to work on the configs based on yur guide |
| 00:04.05 | floren | your* |
| 00:04.08 | ChannelZ | or 192.168.1.8 I mean |
| 00:04.26 | floren | then i will try to do a call and we will know for sure what is good and waht is bad |
| 00:05.14 | floren | ChannelZ: i understood very well your explanation: Your phone sent a totally unauthenticated call to Asterisk which it accepted, sent into the [default] |
| 00:05.17 | floren | <PROTECTED> |
| 00:05.20 | floren | <PROTECTED> |
| 00:05.37 | ChannelZ | but now it doesn't seem to be sending anything anywhere. |
| 00:06.08 | ChannelZ | or it's not getting there, blocked by your router or something, I have no idea. |
| 00:06.11 | floren | my gal now is to clar this puzzle |
| 00:06.28 | floren | i will work on this for the next 30min and if you are still here i will post the results |
| 00:06.32 | ChannelZ | Or your phone and server are both twisted up and networking is confused |
| 00:06.44 | ChannelZ | reboot everything! and smack it. That usually works. |
| 00:06.49 | floren | heh |
| 00:06.59 | floren | i hear you :) |
| 00:07.07 | floren | we do this at work :D |
| 00:07.26 | floren | linux admin for last 15 years |
| 00:08.03 | floren | brb i need to take a break :) |
| 00:09.23 | floren | i think i know why it was working |
| 00:09.34 | floren | i did not used an externip |
| 00:09.38 | floren | now i do |
| 00:09.46 | floren | we will see in a bit, brb |
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| 01:20.57 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 01:58.59 | floren | ChannelZ: still here? |
| 02:16.03 | Katty | evening |
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| 02:24.43 | Grady2000 | heya all, so i'm running 1.6 and when typing: core show channels, I see 11 occurences of an .agi script with state: Up but no one is actually dialing in and using those channels |
| 02:24.55 | Grady2000 | is there any way to tell why they are hung in this state? |
| 02:25.40 | Grady2000 | they are shown as active calls |
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| 03:48.12 | ChannelZ | yawns |
| 03:49.53 | floren | ChannelZ: i found the issue about unspecified device |
| 03:50.21 | floren | how do i disable strict SIP processing in aserisk? |
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| 03:52.57 | ChannelZ | uhh |
| 03:54.14 | floren | http://codepad.org/XK9LzV1M |
| 03:54.29 | floren | this is what is happening when the device tries to register |
| 03:54.40 | floren | i made a basic config with nothing on it |
| 03:54.47 | floren | just for kicks |
| 03:57.24 | floren | http://codepad.org/ZIddrVzr |
| 03:57.28 | ChannelZ | I don't know what all the XML vomit is it's putting in the messages |
| 03:57.39 | floren | :) |
| 03:58.00 | floren | wrong file http://codepad.org/8zQZnv2e |
| 03:58.18 | ChannelZ | Anyway the device does sucessfully register, but then stops responding to Asterisk's acknowledgements by the looks of it. |
| 03:58.27 | floren | ya |
| 03:58.48 | floren | that's why i get an unspecified host, i guess |
| 03:59.02 | ChannelZ | I'd throw your piece of shit phones into the trash and stop trying to make a device which apparently can't follow standards work |
| 03:59.25 | floren | too late heh i paid for it |
| 03:59.32 | ChannelZ | you're paying for sure |
| 03:59.59 | floren | i also got the new NUC to run asterisk on it :S |
| 04:00.18 | floren | at least i will have a sweet device for linux |
| 04:00.51 | floren | http://www.amazon.com/gp/product/B0093LINVK/ |
| 04:01.02 | floren | it's so tiny bob |
| 04:01.11 | floren | perfect asterisk server :) |
| 04:01.36 | carrar | what kind of phone is it? |
| 04:01.51 | floren | cisco 8961, same as the 9971 |
| 04:01.55 | floren | but without camera |
| 04:02.14 | ChannelZ | and all of the 'no-worky' |
| 04:02.21 | carrar | probably should work fine if you can get sip code on it |
| 04:02.22 | floren | heh |
| 04:02.55 | floren | several people on forums got it working, there is a jira thread where they all hover |
| 04:03.17 | floren | iknow this phone in and out now :) |
| 04:03.27 | floren | i just need to get asterisk dance with it |
| 04:04.26 | floren | ChannelZ: i'm going to hook another ip phone to asterisk and see if i can call the "garbage" one :) |
| 04:04.26 | carrar | might try this: http://chan-sccp-b.sourceforge.net |
| 04:05.06 | floren | is a cisco 525g2 :) |
| 04:05.13 | floren | sticking with cisco as you see heh |
| 04:05.16 | carrar | 525 works fine |
| 04:05.20 | floren | ya |
| 04:05.23 | floren | i got it working in 3min |
| 04:05.42 | floren | but 8961 is really sweet |
| 04:05.47 | floren | such a nice phone |
| 04:06.04 | floren | visual voicemail and all this fancy stuff |
| 04:06.34 | floren | i got a good deal for it, paid 200$ |
| 04:06.39 | ChannelZ | All of that crap is built on their own voip server solution |
| 04:06.50 | floren | so that's why i'm fighting with it |
| 04:06.58 | floren | ya |
| 04:07.22 | floren | honestly, i bought only after i saw others managed to fix 9971's |
| 04:07.32 | floren | i said it will be simple, i'm not an idiot |
| 04:07.40 | floren | it looks like is not heh |
| 04:07.52 | floren | but most people use those phones internal |
| 04:08.03 | floren | no voip porviders just their own company |
| 04:08.24 | floren | so i bet a silver dolar it will work fine with teh other cisco phone |
| 04:08.28 | carrar | Cisco phones work great |
| 04:08.32 | carrar | so keep hackin |
| 04:08.49 | floren | bob, are you getting one? :D |
| 04:09.12 | floren | btw, who is david55? |
| 04:09.13 | ChannelZ | but teh 5xx line are totally different beasts than the rest of their SCCP-based poo |
| 04:09.19 | floren | he know a lot of stuff about asterisk |
| 04:10.00 | carrar | Do it right you can order your 525 with your own cert on them and preconfigured for your setup |
| 04:10.05 | floren | i still get xml "vomit" the cli is exploding with the retransmissions |
| 04:10.25 | floren | i already have a 1~525G2 |
| 04:11.10 | floren | i want to use the 525 in the bedroom and the 8961 into living room |
| 04:11.33 | carrar | Thats wise, no one wants to see you in the bedroom |
| 04:11.39 | floren | heh |
| 04:11.59 | ChannelZ | there are much cheaper ways of doing that |
| 04:13.23 | floren | about the retransmissions, someone else had that and david55 told him to "disable strict SIP processing" |
| 04:13.28 | floren | http://forums.asterisk.org/viewtopic.php?p=172524 |
| 04:13.41 | floren | i have no idea how to do that |
| 04:14.20 | floren | pedantic is set to no by default |
| 04:14.31 | floren | i set it anyways and it didnt do jack |
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| 04:14.46 | carrar | pedantic=no |
| 04:14.49 | carrar | in sip.conf |
| 04:14.52 | floren | ya i had that |
| 04:14.57 | floren | no luck |
| 04:15.09 | floren | in asterisk 11 pendantic is default no |
| 04:16.32 | ChannelZ | perhaps there is a "speak SIP correctly" switch in the phone's config |
| 04:16.45 | carrar | heh |
| 04:16.52 | floren | not that i know of :D |
| 04:17.57 | carrar | Are you trying to use the 8961 or 9971 with SIP? |
| 04:18.34 | carrar | probably will have better luck with Chan_SCCP-b |
| 04:18.51 | floren | btw, do i need dahdi? |
| 04:18.58 | carrar | I'd be suprised Cisco even made a SIP version for those phones |
| 04:19.03 | floren | i compiled asterisk with dahdi and libpri |
| 04:19.13 | floren | without* |
| 04:19.30 | floren | i was to lazy to build the rpm's |
| 04:19.43 | floren | so i made it with what centos 6 had in stock |
| 04:20.00 | carrar | oh |
| 04:20.01 | carrar | Cisco Unified IP Phone 8961, 9951, and 9971 (SIP) Release Notes |
| 04:20.03 | carrar | guess so |
| 04:20.21 | floren | for Chan_SCCP-b? |
| 04:20.33 | carrar | ? |
| 04:20.47 | floren | carrar> Cisco Unified IP Phone 8961, 9951, and 9971 (SIP) Release Notes |
| 04:20.58 | carrar | thats Cisco SIP FIrmware release notes |
| 04:21.01 | ChannelZ | you don't need DAHDI |
| 04:21.12 | floren | aha good |
| 04:21.17 | carrar | not sure why you would want to run SIP firmware |
| 04:21.33 | floren | i'm going to build the rpm's anyways this week, just to have a complete solution |
| 04:21.49 | carrar | rpms |
| 04:21.50 | floren | as asterisk is not available in centos, but they included the rules into selinux |
| 04:21.51 | carrar | for what |
| 04:21.54 | floren | go figure |
| 04:22.04 | carrar | Just install Asterisk from source |
| 04:22.08 | floren | nah |
| 04:22.22 | ChannelZ | are you running asterisk on that little black klondike bar computer? |
| 04:22.24 | floren | i build my own rpm's easier and cleaner to maintain |
| 04:22.32 | floren | ya |
| 04:22.34 | floren | heh |
| 04:22.35 | carrar | building RPM's from what |
| 04:22.41 | floren | asterisk |
| 04:22.43 | ChannelZ | then you'll never need DAHDI |
| 04:22.44 | carrar | from Source? |
| 04:23.02 | floren | ya but others might :) |
| 04:23.12 | floren | http://rpm.axivo.com |
| 04:23.18 | ChannelZ | not unless it grows a PCI slot |
| 04:23.27 | floren | especially that is patched with cisco manager |
| 04:23.37 | floren | heh |
| 04:23.46 | floren | i mean for other people bob |
| 04:23.52 | carrar | You should get some Sounds cards in there too |
| 04:24.00 | carrar | make some home made ATA's |
| 04:24.08 | ChannelZ | shrugs |
| 04:24.09 | floren | i have 3 dev servers that i work with |
| 04:24.38 | floren | right now i run asterisk on one of them but i bought the tiny box just for asterisk so i run the tftp server etc |
| 04:24.46 | floren | one tiny thing for everything |
| 04:25.00 | carrar | or get one beefy machine and run KVM |
| 04:25.38 | floren | i have 3 very beefy machines but i need them mostly for my linux stuff |
| 04:25.53 | floren | voip started recently for me |
| 04:26.01 | floren | i never used asterisk until 5 days ago |
| 04:26.03 | carrar | really! |
| 04:26.22 | floren | took me 1 hour to build the rpm's and 5 days to get 3 config lines right in sip.conf :D |
| 04:26.43 | floren | bob and [TK]D-Fender were very helpful today |
| 04:26.54 | floren | they spoon feed me like there is no tomorrow :D |
| 04:27.05 | floren | i was hiding under the table :) |
| 04:29.20 | ChannelZ | not done: http://burner.com/asterisk-primer-beta |
| 04:29.31 | floren | i saw your site bob :) |
| 04:30.43 | floren | This primer is not going to explain every single configuration option or what every single module does |
| 04:30.49 | floren | priceless :D |
| 04:31.15 | floren | asteriskdocs was very useful for me |
| 04:31.36 | floren | i'm going to buy the book just for support |
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| 05:14.29 | floren | ChannelZ: i got the xml issues solved |
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| 05:14.50 | floren | Name/username Host Dyn Forcerport ACL Port Status Description |
| 05:14.53 | floren | 101/101 192.168.1.9 D 5060 Unmonitored |
| 05:14.56 | floren | it shows the ip, finally |
| 05:15.43 | floren | but the retransmit still shows |
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| 12:41.53 | asteriskmonkey | has anyone here been trolled by j2 over patents before? |
| 12:42.15 | WIMPy | j2? |
| 12:42.58 | asteriskmonkey | j2 global, they own efax.com and a bunch of other stuff |
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| 13:38.49 | Synx|hm_ | Anyone familiar with the SIPS URI schema? I'd like to see if Asterisk can enforce the use of SIPS but google isn't helping much. |
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| 14:07.21 | pietro | hello |
| 14:07.46 | pietro | is there a way to instruct Asterisk in order to send a NOTIFY when a peer registers ? |
| 14:07.52 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 14:09.55 | WIMPy | It will happen automatically if you have a hint and the peer was either unregistered or you have qualify enabled. |
| 14:17.47 | pietro | WIMPy: thanks (i miss the qualify). |
| 14:18.59 | pietro | WIMPy: Yes, I can see NOTIFY from Asterisk, but all NOTIFY contain a <state>terminated</state> (for registration and un-registration too) is it correct ? |
| 14:21.12 | WIMPy | I don't usually read SIP debug. |
| 14:21.58 | *** join/#asterisk dwayne (~dwayne@c-71-207-208-112.hsd1.al.comcast.net) |
| 14:22.13 | [TK]D-Fender | NOTIFY's are for presence updates and MWI |
| 14:24.33 | *** join/#asterisk guitarHester (~guitarHes@mobile-166-147-113-111.mycingular.net) |
| 14:25.28 | *** join/#asterisk Thesulac (~Thesulac@82.94.204.46) |
| 14:34.19 | *** join/#asterisk Thesulac (~Thesulac@82.94.204.46) |
| 14:35.25 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.193) |
| 14:44.49 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:45.26 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 14:53.31 | *** join/#asterisk Rahail (~Rahail@67.214.121.181) |
| 14:53.34 | Rahail | hi there |
| 14:53.37 | Rahail | ~pastebin |
| 14:53.37 | 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. |
| 14:54.43 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:54.44 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:56.05 | Rahail | for some reason my asterisk suddenly start acting up |
| 14:56.43 | Rahail | i didnto touch it for last 2 month |
| 14:56.56 | Rahail | http://pastebin.ca/2421349 |
| 14:57.02 | Rahail | how can i get this solv |
| 15:00.56 | [TK]D-Fender | I'd start by updating your *\ |
| 15:01.27 | Rahail | u want me to update it |
| 15:01.27 | Rahail | ? |
| 15:02.25 | [TK]D-Fender | Isn't that what I just said? |
| 15:02.38 | [TK]D-Fender | You version is a few months old... update to get al the current fixes. |
| 15:02.53 | [TK]D-Fender | And if it happen, then we can see if there is something more in play |
| 15:03.00 | [TK]D-Fender | still* |
| 15:09.49 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:11.36 | igcewieling1 | Rahail: turn off debug and those messages should go away |
| 15:12.11 | igcewieling1 | and maybe turning off directmedia |
| 15:15.22 | floren | [TK]D-Fender: i found what was the issue related to my call failures yesterday :) |
| 15:15.40 | floren | i managed to get the phone registered properly with asterisk |
| 15:15.57 | floren | now all i have to do is work on a decent dialing plan :) |
| 15:16.40 | floren | for example i can make inbound calls properly, which i was not able to do that yesterday |
| 15:17.42 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:27.38 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:40.33 | *** join/#asterisk aruntomar (~Thunderbi@49.248.153.139) |
| 15:43.29 | *** part/#asterisk floren (~floren@unaffiliated/floren) |
| 16:02.20 | *** join/#asterisk Cuzner (~ccuzner@198.41.29.45) |
| 16:07.19 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 16:11.06 | *** join/#asterisk tapout (~tapout@unaffiliated/tapout) |
| 16:13.01 | Katty | pokes Qwell |
| 16:13.27 | Katty | checks leifmadsen's flask level |
| 16:13.27 | Qwell | What did I do?! |
| 16:13.32 | Katty | hugs Qwell |
| 16:13.48 | Katty | Qwell: idk, what DID you do? |
| 16:13.54 | Katty | Qwell: all weekend. |
| 16:14.01 | Qwell | ...nothing |
| 16:14.08 | Katty | nothing. or /nothing/ |
| 16:14.19 | Qwell | *whistle*nothing*whistle* |
| 16:14.31 | Katty | ohisee ^_^ |
| 16:17.01 | drmessano | Oh look who it is |
| 16:17.19 | Katty | it's you! |
| 16:18.05 | drmessano | How are you gentlemen. All your base are belong to us |
| 16:18.13 | drmessano | You have no chance to survive. |
| 16:18.16 | drmessano | Make your time. |
| 16:18.18 | drmessano | Ha ha ha ha |
| 16:18.31 | Katty | pats drmessano |
| 16:18.33 | Katty | drmessano: that's nice, dear. |
| 16:18.35 | Katty | eats spaghetti |
| 16:18.53 | Katty | drmessano: what did YOU do all weekend? |
| 16:18.54 | drmessano | Sorry, the "it's you" sent me off |
| 16:19.03 | drmessano | What did *I* do? |
| 16:19.18 | Katty | yes. and what. not who. |
| 16:19.35 | Katty | just to clarify. |
| 16:19.49 | drmessano | I drank, I drove to 7 stores in a 3 mile radius, multiple times, purchasing and exchanging deck furniture, then drank some more. |
| 16:19.50 | drmessano | You? |
| 16:20.33 | Katty | donated some stuff to goodwill. |
| 16:20.45 | Katty | did you ever find something you liked for the deck? |
| 16:23.47 | Cuzner | Lief ain't here right now man... |
| 16:23.52 | Cuzner | Neither is Brad for that matter. |
| 16:24.13 | Cuzner | you know they're up to no good. |
| 16:24.33 | Katty | uh oh. |
| 16:24.54 | Katty | why was i not invited to shenanigans?! :< |
| 16:25.04 | Cuzner | stealth work on opensips or something shady like that i'll bet. |
| 16:25.25 | *** join/#asterisk threesome (~threesome@ip-94-113-12-233.net.upcbroadband.cz) |
| 16:25.36 | *** join/#asterisk Synx|hm (~Synx@unaffiliated/synx-hm/x-1623004) |
| 16:26.19 | drmessano | Katty, yes.. Eventually |
| 16:26.52 | Katty | drmessano: what colors? |
| 16:27.53 | drmessano | Katty, found a few brown plastic adirondack chairs, and a nice brown glass-top table |
| 16:27.57 | beardy | shenanigans and pasta eh. |
| 16:28.01 | drmessano | The issue was finding comfy chairs |
| 16:28.20 | igcewieling1 | In case anyone feels like testing it http://140.239.234.172/transcode/transcode.php |
| 16:28.31 | drmessano | My bald head hates hitting the back of a chair with a curve in the top. I needed something with a straight back |
| 16:28.55 | Katty | drmessano: finding patio furniture sure is a pain. i'm still trying to find a set myself. |
| 16:29.01 | Katty | drmessano: with rocking chairs. |
| 16:30.09 | *** join/#asterisk sawgood (~sawgood@23-24-214-61-static.hfc.comcastbusiness.net) |
| 16:31.32 | drmessano | Ahhh |
| 16:40.47 | Weezey | Why is it that rtp.conf seems to be ignored in asterisk 11 ? |
| 16:41.54 | Weezey | Probation passed - setting RTP source address to 192.168.1.241;25656 |
| 16:42.12 | igcewieling1 | Weezey: 192.168.1.241 is the address of your server? |
| 16:42.14 | Weezey | rtp.conf has start and end set to 10000 to 20000 |
| 16:42.15 | Weezey | no |
| 16:42.21 | Qwell | Weezey: because you don't understand networking |
| 16:42.26 | Qwell | source port != destination port |
| 16:44.43 | Weezey | okay, so where do I see the destination port? |
| 16:45.03 | Qwell | in the SDP, or in a packet capture, or an rtp debug |
| 16:45.16 | Weezey | I do rtp debug and I see packets from end point 1 on port 6050 and end point 2 27512 |
| 16:45.27 | Weezey | both outside the range set in rtp.conf |
| 16:46.07 | *** join/#asterisk paulc (~root@unaffiliated/paulc) |
| 16:46.25 | igcewieling1 | Weezey: show us with a pastebin |
| 16:46.55 | igcewieling1 | Weezey: the ASTERISK side will use the ports specified in rtp.conf the client ports are assigned by the client. |
| 16:47.12 | *** join/#asterisk vlad_starkov (~vlad_star@87.117.139.75) |
| 16:48.05 | Weezey | maybe it's a direct media problem them. |
| 16:48.07 | Weezey | n |
| 16:48.35 | Weezey | I'm seeing it everywhere, so it must be something that's being ignored in my config from 1.8 to 11 |
| 16:48.40 | Weezey | random 1 way audio |
| 16:49.20 | igcewieling1 | *nod* We get random one-way audio with re-invites enabled too. We turned them off and decided to figure it out later. No NAT involved, of course. |
| 16:49.48 | igcewieling1 | because of contractual limitations with one of our carriers (the one used most) we MUST NOT reinvite many of our calls. |
| 16:50.28 | WIMPy | Because you must not irritate the NSA? |
| 16:50.52 | Weezey | yeah, I had it off completely, always |
| 16:51.09 | Weezey | directmedia=no ? |
| 16:51.21 | igcewieling1 | WIMPy: because if we re-invited, the data would be considered "transit" and we are not allowed to use that connection for transit. |
| 16:53.10 | igcewieling1 | Weezey: directmedia=no in [general] and no peers have a directmedia option? |
| 16:54.07 | Weezey | they are from the canreinvite days in most cases. |
| 17:03.48 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 17:03.48 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 17:05.29 | igcewieling1 | canreinvite=yes enables direct media |
| 17:06.19 | [TK]D-Fender | that's "directmedia" in 1.6+ |
| 17:07.59 | igcewieling1 | [TK]D-Fender: right, but does asterisk 11 IGNORE canreinvite? |
| 17:08.26 | [TK]D-Fender | I think so... |
| 17:08.28 | igcewieling1 | (12:53:09 PM) igcewieling1: Weezey: directmedia=no in [general] and no peers have a directmedia option? |
| 17:08.28 | igcewieling1 | (12:54:07 PM) Weezey: they are from the canreinvite days in most cases. |
| 17:08.49 | igcewieling1 | still best to remove them just in case. |
| 17:17.36 | *** join/#asterisk geeksteve (~geeksteve@195.110.168.123) |
| 17:20.12 | *** join/#asterisk peetaur2 (~peter@x2f0d09f.dyn.telefonica.de) |
| 17:20.13 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 17:26.55 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 17:39.17 | leifmadsen | Katty: FLASK LEVEL CRITICAL! |
| 17:44.25 | *** join/#asterisk danyasd (d4ff17d0@gateway/web/freenode/ip.212.255.23.208) |
| 17:45.47 | *** join/#asterisk peetaur2 (~peter@x2f0d09f.dyn.telefonica.de) |
| 17:46.43 | *** join/#asterisk MLNoah (~chatzilla@noc.metalink.net) |
| 17:47.10 | MLNoah | can anyone point me in the direction of a resource for configuring Asterisk to use 1 d-channel across 2 PRI connections? |
| 17:50.29 | pabelanger | nfas |
| 17:53.09 | *** join/#asterisk peter (~peter@x2f0d09f.dyn.telefonica.de) |
| 17:53.18 | MLNoah | thank you |
| 17:56.23 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:56.24 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:56.27 | danyasd | Hey, I have a problem with asterisk on my raspi. i followed a tutorial and in the very beginning, the command console dial xx is used. the cli says theres no such command. i think its because of the chan_alsa.so module, asterisk can't load that. I get this error: http://f.666kb.com/i/cfmbowigpmvmhgo9c.gif Can anyone help me please? |
| 17:58.04 | Katty | refills leifmadsen's flask |
| 17:58.09 | *** join/#asterisk robl^ (robl@pdpc/supporter/active/robl) |
| 17:58.16 | Katty | leifmadsen: there there. everything will be ok. two shots and a nap for you! |
| 17:58.26 | leifmadsen | WHO HAS TIME TO NAP?! |
| 17:58.31 | leifmadsen | has a third shot |
| 17:58.40 | Katty | better make that 4. |
| 17:58.54 | Katty | you know what, let me get some everclear for you instead. |
| 18:00.32 | [TK]D-Fender | danyasd: If you don't have OSS/ALSA... then you have no console to dial with. Use a softphone, etc on ANOTHER PC to be your "phone: |
| 18:00.41 | [TK]D-Fender | danyasd: * CLI makes a crappy phone. |
| 18:01.03 | Katty | hello fender bender |
| 18:01.40 | [TK]D-Fender | Katty: Mew. |
| 18:01.51 | danyasd | [TK]D-Fender : thanks, so it has no impact on the functionality of asterisk itself, except the CLI ? |
| 18:01.58 | Katty | [TK]D-Fender: how're you dear? did you have a nice weekend |
| 18:02.12 | [TK]D-Fender | danyasd: That's all that module is good for really... |
| 18:02.33 | [TK]D-Fender | Katty: Rain FUBAR'd my getting out on the lake :( |
| 18:02.47 | [TK]D-Fender | Katty: This has been a really crappy summer climate-wise so far |
| 18:03.32 | Katty | [TK]D-Fender: i'm sorry to hear that )= did you at least get all your band promo stuff going? |
| 18:03.43 | Katty | [TK]D-Fender: or maybe you weren't doing that this weekend, and just met about it. my memory is a bit fuzzy |
| 18:04.02 | [TK]D-Fender | Katty: They see me posting... they hatin'! |
| 18:04.16 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-xexiacvesvxxyixe) |
| 18:04.24 | [TK]D-Fender | Katty: WWorking the "buzz" up... |
| 18:04.48 | [TK]D-Fender | Katty: And my other group seems to have just gotten our previous guitarist back which is absolutely awesome... |
| 18:04.51 | Katty | [TK]D-Fender: that must be a marketing term i'm not familiar with |
| 18:05.53 | *** join/#asterisk GameGamer43 (uid5533@gateway/web/irccloud.com/x-cpranlfeerrxiquz) |
| 18:08.00 | [TK]D-Fender | Katty: "Buzz"? Can't see how that'd be new... |
| 18:14.49 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:14.49 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:19.55 | jeev | hm, was hoping the new asterisk logging would give me the IP of Received incoming SIP connection from unknown peer to |
| 18:21.22 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.189) |
| 18:24.46 | *** join/#asterisk vfabi (~fabi@host-static-37-75-109-141.moldtelecom.md) |
| 18:26.13 | jeev | oh hm, i only did security |
| 18:31.41 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.189) |
| 18:36.32 | asteriskmonkey | Does rusty hang out on IRC? |
| 18:37.41 | asteriskmonkey | is there anyone fro digium currently on in this room/ |
| 18:38.20 | igcewieling1 | ~ask |
| 18:38.20 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 18:39.09 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 18:39.53 | *** join/#asterisk catphish (~J@unaffiliated/catphish) |
| 18:44.03 | catphish | does anyone have a PSTN number i can call with an echo test, endless loop, or other endless junk? |
| 18:50.54 | *** join/#asterisk Defraz (~Defraz@24-116-129-18.cpe.cableone.net) |
| 18:55.50 | *** join/#asterisk aelliott (~aelliott@198.41.29.45) |
| 18:57.35 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 19:00.01 | paulc | catphish: where at? I could probably knock something up if needed |
| 19:02.37 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 19:03.32 | catphish | paulc: i'm in the UK, but it doesn't matter where it is |
| 19:03.38 | catphish | as long as it's a sanely priced place |
| 19:03.52 | catphish | probably don't need it until tomorrow now |
| 19:04.08 | paulc | I've got an 0874 number in the UK I could point somewhere temporarily.. or a bunch of US/Canada numbers I can muck around with for testing |
| 19:04.18 | paulc | is that British bloke living in Vancouver BC |
| 19:04.41 | catphish | 087 = expensive :( |
| 19:04.54 | catphish | don't go to too much trouble |
| 19:04.55 | paulc | If it's just for making test calls I can give you a chatline type number that's good for a laugh.. or if you explicitly want echotest etc, that's doable too |
| 19:05.07 | catphish | lol |
| 19:05.37 | catphish | i just want something non-premium that will keep a call open as long as i leave it connected |
| 19:05.51 | catphish | need to test some call transfer stuff |
| 19:06.26 | catphish | i need to investigate what happens to transferred calls in the CDR |
| 19:09.20 | ChannelZ | has to be PSTN? |
| 19:13.59 | Chainsaw | I was able to provide a local non-premium number. |
| 19:17.52 | igcewieling1 | for the most part long distance companies do echo canceling, so calling long distance isn't much of an echo test |
| 19:18.12 | igcewieling1 | at least if you are trying to resolve echo issues |
| 19:18.47 | Chainsaw | It's about CDR updating & billing during transfers. |
| 19:18.55 | catphish | echo test is different from "echo" |
| 19:19.02 | Chainsaw | So you need a test call that sticks around as you keep changing the A end. |
| 19:19.03 | catphish | echo test is for measuring latency |
| 19:19.33 | catphish | so you speak and it deliberately echoes back to you |
| 19:19.54 | catphish | but as Chainsaw says, i just wanted a number that would bill me consistently |
| 19:20.17 | catphish | while i test transfers at my end |
| 19:20.55 | Chainsaw | And does non-CCM hold music, because everyone knows that by heart. Not in a good way. |
| 19:21.28 | catphish | lol |
| 19:27.23 | *** join/#asterisk zerohalo (~zerohalo@c-174-62-142-83.hsd1.ma.comcast.net) |
| 19:41.49 | *** join/#asterisk Defraz (~Defraz@24-116-129-18.cpe.cableone.net) |
| 19:59.35 | *** part/#asterisk catphish (~J@unaffiliated/catphish) |
| 20:04.22 | *** join/#asterisk asteriskmonkey (~Tardis@74-51-38-204.telnetcommunications.com) |
| 20:05.45 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 20:16.03 | *** join/#asterisk tilt_ (~tilt@173-13-180-97-sfba.hfc.comcastbusiness.net) |
| 20:16.09 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 20:16.12 | asteriskmonkey | im runnning asterisk 10.5 , getting a wierd issue, rebooting it dosnt seem to solve it |
| 20:16.18 | asteriskmonkey | ast_openstream_full: File /shared/asterisk/sounds/custom/ivr does not exist in any format |
| 20:16.21 | asteriskmonkey | i see the file |
| 20:16.25 | asteriskmonkey | perms are set good |
| 20:16.34 | asteriskmonkey | :/ dont know where to debug from here |
| 20:17.30 | Qwell | asteriskmonkey: logs? |
| 20:17.33 | igcewieling1 | asteriskmonkey try su asterisk "ls -l /shared/asterisk/sounds/custom/ivr* |
| 20:18.50 | asteriskmonkey | thanks |
| 20:18.58 | asteriskmonkey | perms are like 777 and i get perm deniied |
| 20:18.58 | asteriskmonkey | wierd |
| 20:19.18 | Qwell | Do you have access to every dir leading up to it? Is the dir scannable? |
| 20:19.39 | asteriskmonkey | yeah |
| 20:19.44 | Qwell | prove it |
| 20:19.53 | asteriskmonkey | im doing it from cli |
| 20:19.57 | asteriskmonkey | as asterisk |
| 20:19.58 | asteriskmonkey | :/ |
| 20:20.05 | asteriskmonkey | ill chown it see if that sorts it |
| 20:21.19 | Qwell | show logs |
| 20:21.50 | asteriskmonkey | ^ chowning it to asterisk works |
| 20:21.55 | asteriskmonkey | but should have worked other way too |
| 20:22.01 | asteriskmonkey | weirdness.. |
| 20:24.21 | *** join/#asterisk heffer (felix@fedora/heffer) |
| 20:25.47 | *** join/#asterisk kresp0 (~kresp0@30.Red-88-6-105.staticIP.rima-tde.net) |
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| 20:32.35 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
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| 20:40.48 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 20:40.48 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 20:41.54 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 20:47.14 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 21:02.27 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 21:13.33 | *** join/#asterisk apb1963 (~apb1963@174.134.117.244) |
| 21:14.37 | *** join/#asterisk DarthExpeditor (~zbriggs@rrcs-71-43-76-226.se.biz.rr.com) |
| 21:14.48 | DarthExpeditor | anyone here use Vitelity? |
| 21:15.21 | apb1963 | Is that like Viagra? |
| 21:15.27 | apb1963 | j/k |
| 21:16.06 | Qwell | ~ask |
| 21:16.06 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 21:17.42 | igcewieling1 | We have users reporting lack of ringback when calling some numbers. All SIP. Does anyone have any pointers on diagnosing and fixing? |
| 21:18.00 | igcewieling1 | Asterisk 1.4 and Asterisk 11 involved |
| 21:18.14 | drmessano | Vitelity is like viagra. It's good when it stays up, but it's never up for more than 4 hours |
| 21:18.24 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 21:22.30 | DarthExpeditor | LOL |
| 21:22.33 | DarthExpeditor | I like that. |
| 21:22.35 | DarthExpeditor | That's funny |
| 21:23.06 | DarthExpeditor | We use it for 800 numbers |
| 21:23.32 | DarthExpeditor | I was just going to see if anyone else was having any issues but then realized that the PBX locked up. |
| 21:27.10 | igcewieling1 | I just can't figure out prematuremedia and progressinband |
| 21:29.36 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:31.21 | *** part/#asterisk aelliott (~aelliott@198.41.29.45) |
| 21:34.44 | *** part/#asterisk DarthExpeditor (~zbriggs@rrcs-71-43-76-226.se.biz.rr.com) |
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| 22:06.44 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 22:11.30 | *** join/#asterisk rasa_ (~rasa@178.206.117.93) |
| 22:11.36 | *** join/#asterisk floren (~floren@unaffiliated/floren) |
| 22:11.43 | rasa_ | hello people! |
| 22:11.50 | rasa_ | Here is anybody? |
| 22:12.34 | rasa_ | I have trouble with area code |
| 22:13.13 | [TK]D-Fender | Which is? |
| 22:13.29 | rasa_ | <PROTECTED> |
| 22:13.51 | rasa_ | I install FreePBX w/*1.8 and can't call to any my city phonenumber. I add in Outbound Route template full area code (prepend) of my city and after that I can call. |
| 22:14.25 | floren | ChannelZ: your site links are gone? |
| 22:14.35 | floren | www.burner.com/asterisk-primer-beta/configuring-sip/ |
| 22:14.50 | floren | hi [TK]D-Fender |
| 22:14.51 | rasa_ | Then I install Elastix (*1.8 too) and all calls to my city phonenumbers go without any area codes. WOW, thats very good told I. And in Elastix Outbound Route I has only dot in match pattern. |
| 22:15.11 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 22:15.12 | [TK]D-Fender | rasa_: from-internal and from-trunk are COMPLETELY different. You should not be referencing them both in this |
| 22:15.15 | rasa_ | hmm your link wanna sign in? |
| 22:15.32 | floren | ya, is ChannelZ' site |
| 22:15.40 | floren | for some reason is asking me to sign in |
| 22:16.01 | [TK]D-Fender | rasa_: join #freepbx and continue in there |
| 22:16.10 | [TK]D-Fender | rasa_: This is a FreePBX issue, not an Asterisk issue |
| 22:16.20 | [TK]D-Fender | rasa_: And should be continued in there |
| 22:16.24 | rasa_ | [TK]D-Fender: thank you |
| 22:16.34 | rasa_ | [TK]D-Fender: but one thing is intersting |
| 22:16.56 | [TK]D-Fender | context=from-trunk-sip-outgoing-Trunk <- you should never ben having to specify a context like tis |
| 22:17.02 | [TK]D-Fender | be* |
| 22:17.35 | rasa_ | [TK]D-Fender: why not? security reason? |
| 22:17.42 | floren | [TK]D-Fender: let me know if you are busy |
| 22:17.57 | rasa_ | thats automatically generated from Elastix |
| 22:18.01 | [TK]D-Fender | rasa_: Because you are specifying separate places and not letting the system's logic do what it has to do |
| 22:18.17 | [TK]D-Fender | Routes have a prioirty and when you point things to one particular part of a chain you break that order |
| 22:20.13 | rasa_ | Sure, I'll trying to join in #freepbx, but I think is more asterisk admins knowledge - how it can be - elastix has outgoing trunk as default |
| 22:20.27 | rasa_ | and freepbx has outgoing route as default |
| 22:20.48 | rasa_ | AND WHY I SHOULDNT enter area code |
| 22:20.54 | rasa_ | (is most interesting) |
| 22:21.40 | rasa_ | if I abstract from any web-gui's can you tell me - should I enter local area or not? |
| 22:22.50 | [TK]D-Fender | You should enter whatever you WANT to enter... or have TO ENTER |
| 22:23.06 | [TK]D-Fender | Should you? |
| 22:23.07 | [TK]D-Fender | I don't know |
| 22:23.17 | [TK]D-Fender | What are your users USED to entering? |
| 22:23.27 | [TK]D-Fender | How does your trunk demand you to format your numbers? |
| 22:23.40 | [TK]D-Fender | In the end you have to give the trunk something it will accept. |
| 22:23.55 | rasa_ | My trunk has fromuser=336556 string |
| 22:24.17 | rasa_ | (my city has 6-digits numbers) |
| 22:24.30 | [TK]D-Fender | That has nothing to do with "aer codes", or routes. |
| 22:24.33 | [TK]D-Fender | area* |
| 22:24.54 | rasa_ | Nothing to do? But.. |
| 22:25.18 | [TK]D-Fender | no "but"... that is part of how you are forcing an identity. It has nothing to do with a number you dial |
| 22:25.38 | [TK]D-Fender | it sets the FROM header of a dial.... not the destination you are actually dialing |
| 22:25.53 | [TK]D-Fender | BRB |
| 22:27.56 | rasa_ | [TK]D-Fender: with all my health I try to find the method return back "Elastix-without-any-area-codes" call method |
| 22:28.24 | rasa_ | I try understand - why two same GUI call different |
| 22:29.20 | rasa_ | One can dial without definite locak area code (via trunk directly) and second (FreePBX) - can't |
| 22:29.34 | rasa_ | (trough one of outbound route) |
| 22:30.38 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 22:31.49 | rasa_ | [TK]D-Fender: you tell, context from-trunk-sip-outgoing-Trunk is very wrong, but it dial any city number without any local area codes |
| 22:32.14 | rasa_ | this is so strange |
| 22:33.11 | rasa_ | If manual move my extension (namba 777, usual ATA-phone) to context from-trunk-sip-outgoing-Trunk then I again make calls to city without any local area code. |
| 22:35.52 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.252) |
| 22:35.52 | *** join/#asterisk g_r_eek (~g_r_eek@78-34-183.adsl.cyta.gr) |
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| 22:36.17 | [TK]D-Fender | rasa_: You should not be referncing anything other than from-internal for your extensions, and trom-trunk for your trunks |
| 22:36.45 | [TK]D-Fender | your OUTBOUND ROUTES should be doing all of the manipulation of what you want the USER to dial and transform it into the format the TRUNK requires |
| 22:39.06 | rasa_ | [TK]D-Fender: sounds like I should mandatory are codes in my outbound route? :( |
| 22:39.24 | rasa_ | are=*area |
| 22:39.45 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 22:40.10 | [TK]D-Fender | You don't seem sure of what your trunk even requires.... |
| 22:40.27 | [TK]D-Fender | Route matches what you expect the USER to dial |
| 22:40.47 | [TK]D-Fender | And then ADD things or REMOVES them as needed and then passes that on to the trunk |
| 22:41.24 | rasa_ | I know what wanna my users - they wanna dial 6-digits numbers wihout any local area code |
| 22:41.27 | floren | [TK]D-Fender: you have a lot of patience and willing to help, i apreciated that TK |
| 22:41.36 | floren | i was reading what you posted |
| 22:42.37 | rasa_ | floren: you about me? Yes I am noob |
| 22:42.59 | floren | i'm a noob too, and he helped me yesterday |
| 22:43.09 | floren | looks like he helps everyone :) |
| 22:43.28 | rasa_ | I am trying understand how this F////ing trunk outgoing can dial WITHOUT area code |
| 22:44.23 | [TK]D-Fender | [18:41]rasa_I know what wanna my users - they wanna dial 6-digits numbers wihout any local area code <- then do that |
| 22:45.12 | [TK]D-Fender | rasa_: If your trunk requires the area code... THEN ADD IT ON <- |
| 22:45.21 | [TK]D-Fender | rasa_: that is what the PREPEND field is for |
| 22:45.26 | rasa_ | [TK]D-Fender: i DID that - with adding prepend with my area code, |
| 22:45.38 | rasa_ | but it seems suspicious for me |
| 22:45.58 | [TK]D-Fender | How is it "suspicious"? |
| 22:46.58 | rasa_ | why the trunk in default install of elastix can dial to city WITHOUT any area code. My brain is bricked with that |
| 22:47.39 | [TK]D-Fender | There is no such thing as default |
| 22:47.49 | [TK]D-Fender | Your outbound routes are the rules. |
| 22:48.00 | [TK]D-Fender | You TRUNK can also modify what your outbound routes give them |
| 22:48.04 | [TK]D-Fender | This is all YOUR job |
| 22:48.08 | [TK]D-Fender | Forget "default" |
| 22:48.28 | [TK]D-Fender | You need to know what you want the user to dial, and what the trunk expects an make sure to modify A into B |
| 22:48.58 | rasa_ | [TK]D-Fender: I understand you perfectly |
| 22:50.01 | rasa_ | Okay this f/////g outbound routes can't dial without area code (when trunk outgoing CAN). |
| 22:50.16 | rasa_ | so my specially way is only way - |
| 22:50.36 | rasa_ | using local area in outbound route in anyway |
| 22:51.07 | [TK]D-Fender | You're doing it wrong. |
| 22:51.30 | rasa_ | and just use . (dot) as match patterin in trunk outgoing |
| 22:51.51 | [TK]D-Fender | the outbound route doesn't dial. It only matches what it is TOLD to match, manipulates it the way it is told to (prepend / prefix), and then PASSES that on to the trunk |
| 22:52.03 | [TK]D-Fender | You have made yours in a way that does not AGREE |
| 22:53.03 | rasa_ | ooh, I understood. Outbound route - is just a filter |
| 22:53.43 | rasa_ | or rules manipulator |
| 22:53.47 | [TK]D-Fender | Correct |
| 22:53.55 | [TK]D-Fender | the trunk can ALSO manipulate |
| 22:54.03 | rasa_ | yes I saw |
| 22:54.09 | rasa_ | same look like |
| 22:54.13 | rasa_ | match patterns |
| 22:55.52 | rasa_ | [TK]D-Fender: so I have big global question |
| 22:56.52 | rasa_ | [TK]D-Fender: Is normal to call to your living city, to your citizens, to your neighborhood home USING local area code in pattern? |
| 22:57.04 | [TK]D-Fender | yes |
| 22:57.09 | [TK]D-Fender | thqat is 10-digit dialing here |
| 22:57.14 | [TK]D-Fender | I chose NOT to force this |
| 22:57.20 | rasa_ | I am from Russia |
| 22:57.50 | [TK]D-Fender | So if my users dial the old 7-digit number my system ASSUMES I mean a LOCAL number and it adds my LOCAL area-code so that the over number dialed is 10-digits like my telco expects |
| 22:59.05 | rasa_ | bingo! your system behaivior as my the from-trunk context |
| 22:59.25 | rasa_ | it self give prepend |
| 22:59.25 | *** join/#asterisk cmendes0101| (~cmendes01@72.1.46.254) |
| 23:00.07 | apb1963 | gives himself a prepend |
| 23:00.29 | rasa_ | "sorry for my terrible" |
| 23:00.41 | apb1963 | +1apb1963 |
| 23:00.46 | rasa_ | English not my native |
| 23:00.50 | rasa_ | You're doing it wrong. |
| 23:00.57 | [TK]D-Fender | No... |
| 23:00.58 | rasa_ | omm |
| 23:01.01 | rasa_ | no no |
| 23:01.04 | [TK]D-Fender | from-trunk is where calls FROM your trunk go |
| 23:01.09 | [TK]D-Fender | this has NOTHING to do with DIALING OUT |
| 23:01.11 | rasa_ | it was quote |
| 23:03.46 | rasa_ | [TK]D-Fender: does it mean trunk stupidly PASSE data via himself? |
| 23:04.36 | cmendes0101| | I'm trying to find a solution for adding a recording beep. I see the dial() has a option L but is there anyway to stop that beeping mid call? we have a feature code that can start and stop recording so trying to have those match up |
| 23:05.27 | rasa_ | cmendes0101|: "recording beep"? mandatory of law? |
| 23:05.51 | [TK]D-Fender | rasa_: No it means you are mixing up IN vs OUT in your head constantly |
| 23:06.00 | cmendes0101| | yah. Just playing beep every x seconds |
| 23:06.26 | *** join/#asterisk DEMNVT (~Adium@rmsaus7.lnk.telstra.net) |
| 23:06.51 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 23:06.55 | [TK]D-Fender | rasa_: "from-trunk" has nothing to do with dialing out. |
| 23:08.17 | cmendes0101| | In the doc it says theres a set of variables for the L option like LIMIT_PLAYAUDIO_CALLEE. If I update the variable mid call will that do anything? |
| 23:08.49 | rasa_ | [TK]D-Fender: The stupid filter "Outbound route" passed data to "from-trunk" which make nothing do with dialing out. Nothing doing nothing. My head explose now. |
| 23:11.13 | [TK]D-Fender | The stupid filter "Outbound route" passed data to "from-trunk" <- no |
| 23:11.21 | [TK]D-Fender | outbound route's do NOT point there |
| 23:11.31 | [TK]D-Fender | YYou are STILL backwards |
| 23:16.26 | igcewieling1 | sounds like freepbx |
| 23:16.59 | tm1000 | explose` s |
| 23:17.23 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 23:18.09 | rasa_ | [TK]D-Fender: jooooj, here is 3 am :( |
| 23:18.22 | [TK]D-Fender | igcewieling1: it is... and I've got this :) |
| 23:18.30 | tm1000 | igcewieling1: you didnt hear. this is ##freepbx |
| 23:18.37 | rasa_ | [TK]D-Fender: thank you for ur patience |
| 23:18.51 | igcewieling1 | [TK]D-Fender: I wasn't intending on helping. 8-| |
| 23:19.17 | rasa_ | [TK]D-Fender: have you any contact - jabber, skype, icq or smth other? |
| 23:19.38 | [TK]D-Fender | rasa_: yes, and no intention of using any of them for this |
| 23:21.12 | rasa_ | [TK]D-Fender: da may be you can give me any of them? ) |
| 23:21.46 | tm1000 | rasa_: did you miss what he just said? |
| 23:21.53 | tm1000 | "no intention of using any of them for this" |
| 23:22.50 | rasa_ | tm1000: yes, u r right. just my time is 3 am. going to sleep. I can talking to [TK]D-Fender only at my nights |
| 23:23.07 | [TK]D-Fender | rasa_: the "context=" you specify in your trunk is for what calls FROM that trunk go. This should almost always be "from-trunk. Those calls lead to INBOUND ROUTES. |
| 23:23.23 | [TK]D-Fender | rasa_: Your trunk's context has NOTHNG to do with what your PHONE dials |
| 23:23.58 | [TK]D-Fender | rasa_: Your PHONE's context should be "from-internal". That will let it go through the chain of your INBOUND ROUTES and in that process choose which trunks to use to dial out. |
| 23:24.51 | rasa_ | [TK]D-Fender: know I should tell that in #freepbx, but: |
| 23:25.12 | [TK]D-Fender | wondws why there is always a "but" |
| 23:25.15 | [TK]D-Fender | wonders* |
| 23:25.37 | rasa_ | [TK]D-Fender: : )) |
| 23:26.36 | *** join/#asterisk Synx|hm_ (~Synx@unaffiliated/synx-hm/x-1623004) |
| 23:27.04 | rasa_ | now of all of my extensions works via "from-internal", not from elastix's default "from-trunk" |
| 23:28.09 | rasa_ | u gave me answer to big question - be there from local area code? YES. Okay I'll use it. |
| 23:29.08 | [TK]D-Fender | [19:26]rasa_now of all of my extensions works via "from-internal", not from elastix's default "from-trunk"<- this is wrong |
| 23:29.32 | rasa_ | f//////ck man how is wrong!!! |
| 23:29.35 | [TK]D-Fender | Extensions default to "from-internal" |
| 23:29.41 | rasa_ | you told vice versa |
| 23:29.47 | [TK]D-Fender | ... |
| 23:29.50 | [TK]D-Fender | No, I didn't |
| 23:29.53 | [TK]D-Fender | you have a reading problem' |
| 23:30.04 | [TK]D-Fender | [19:23][TK]D-Fenderrasa_: Your PHONE's context should be "from-internal". That will let it go through the chain of your INBOUND ROUTES and in that process choose which trunks to use to dial out. |
| 23:30.14 | [TK]D-Fender | Phone = from-internal |
| 23:30.18 | [TK]D-Fender | phone = extension. |
| 23:30.22 | [TK]D-Fender | extension = from-internal |
| 23:30.27 | [TK]D-Fender | NOT "from-trunk" |
| 23:31.54 | rasa_ | yes!so I wrote to you: [19:26]Irasa_ not from elastix's default "from-trunk" |
| 23:32.12 | rasa_ | "NOT FROM" |
| 23:33.43 | rasa_ | give god the health to you, I go to sleep |
| 23:34.16 | [TK]D-Fender | Come back fresh when you're ready to read things in-context. |
| 23:34.33 | [TK]D-Fender | sleep = good |
| 23:34.56 | rasa_ | When I woke up u'll sleeping :) |
| 23:36.10 | rasa_ | Thank you for your explains, good day and good night |
| 23:36.35 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.119) |
| 23:36.54 | floren | [TK]D-Fender: got time to look at a sip trace? |
| 23:37.56 | *** join/#asterisk TJNII (~TJNII@75-1-144-147.lightspeed.snantx.sbcglobal.net) |
| 23:39.12 | [TK]D-Fender | floren: Share it anyway. Don't wait for one source of help. |
| 23:39.17 | [TK]D-Fender | Show to all, hope for ANY |
| 23:40.09 | floren | http://codepad.org/MuCQxXew |
| 23:40.25 | floren | i managed to get the phone stright yesterday, it registers properly |
| 23:40.54 | floren | but i dont understand why the reregistrations, pedantic is set to no by default in asterisk |
| 23:42.43 | floren | anyways, the phone works for outbound calls, with the config we worked yesterday |
| 23:43.05 | floren | i still have to work the dian plan as right now when i ring my DID, it answers automatically for me :) |
| 23:43.07 | [TK]D-Fender | BRB, lanudry |
| 23:43.11 | [TK]D-Fender | laundry |
| 23:43.26 | floren | have fun :) |
| 23:44.21 | floren | i was dreaming when i thought it will be easy to setup asterisk. can't do much when you dont know a product :) |
| 23:54.08 | [TK]D-Fender | Asterisk is easy |
| 23:54.24 | [TK]D-Fender | You are the one who brough CCM and a busted SIP implementation and proxying into this... |
