| 00:00.11 | saint_ | so Asterisk can have the status of the line ? |
| 00:00.49 | WIMPy | din3sh: Not neccessarily. You should read a lot about load balancing / LVS and HA. |
| 00:01.20 | WIMPy | saint_: Hintes are for devices and some special other states, not for extensions. |
| 00:01.40 | saint_ | WIMPy: so I have a couple of digium phones. |
| 00:01.46 | WIMPy | It's an extension->state mapping. |
| 00:01.52 | saint_ | WIMPy: do i need to put a hint for every exten => when calling htem ? |
| 00:02.12 | din3sh | ok |
| 00:02.20 | WIMPy | IIRC hintes are created automatically when you use DPMA. |
| 00:02.58 | rue_work | wow, that was a cool problem, a wrong dial plan on an aastra set caused it not to connect the incomming audio stream |
| 00:03.22 | saint_ | WIMPy: ok... |
| 00:03.51 | navaismo | <PROTECTED> |
| 00:07.56 | saint_ | does SLA work with SIP ? Or only Analog / T1 trunks ? |
| 00:09.15 | saint_ | ha.. never mind, i just found the example with sip trunks |
| 00:09.55 | pabelanger | din3sh: well, if you add a 3rd server, you now need to make sure it is redundant too |
| 00:10.04 | pabelanger | otherwise, it is a single point of failure |
| 00:10.30 | pabelanger | Do active / passive, have your phone reregister every 60 seconds |
| 00:10.41 | saint_ | I have meetme.so , but I do not have SLA. What did I do wrong ? The documentation says that if you have meetme, then SLA should be loaded too. |
| 00:10.48 | pabelanger | if server dies, phone will reregister in 60 seconds |
| 00:15.09 | din3sh | @pabelanger: ok thanks |
| 00:17.56 | pbxbrian | Running Asterisk 11 since mid November without any issue.. The quality of initial releases has got so much better. |
| 00:18.40 | *** join/#asterisk Defraz (~Defraz@24-116-129-18.cpe.cableone.net) |
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| 00:30.08 | leifmadsen | saint_: that's true -- SLA is included in the meetme module |
| 00:30.29 | leifmadsen | saint_: btw the documention in the 3rd edition of Asterisk is kind of wrong |
| 00:30.39 | leifmadsen | use ofps.oreilly.com and read the 4th edition where I updated it |
| 00:30.48 | leifmadsen | I think that is the only place with sane documentation for the SLA functionality |
| 00:31.01 | leifmadsen | please test and review what we have there and provide feedback tonight if you can |
| 00:31.09 | leifmadsen | runs away again |
| 00:36.23 | SeRi | does any body know where I can ppurchase a Polycom Productivity Suite Package? |
| 00:36.35 | SeRi | or individual license? |
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| 00:38.51 | ChannelZ | I'm guessing only from them. |
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| 01:03.37 | rue_house | ChannelZ, yes, replaced lots of roofs, today your price is $7000 |
| 01:06.41 | SeRi | ChannelZ: nope |
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| 01:27.15 | din3sh | zzzZZZzzzZZZzzz |
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| 03:07.44 | zhando | need pointers to get a soundcard channel working alsa/oss/console whatever's easiest... I'm on an hp laptop.. |
| 03:12.37 | zhando | I mean I fire up vlc and I hear sound no problem.. Why is it so hard with * ??? |
| 03:16.25 | zhando | I'm on linux too |
| 03:24.22 | saint_ | hey, I know I have the right login and password since I did a copy / paste in jabber.conf. |
| 03:24.32 | saint_ | But when I start my asterisk box, I get an error message: JABBER: encryption failure. possible bad password. |
| 03:24.51 | saint_ | Anyone would have a trick on how to trace or fix that ? |
| 03:24.58 | saint_ | I'm 100% sure of the login and password |
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| 03:32.03 | saint_ | i wonder if it's because my google voice is not a gmail.com account .. |
| 03:38.25 | saint_ | anyone can give me a hand with this http://pastebin.com/GtGapqYy |
| 03:54.49 | saint_ | ha yeah, i confirm, it's a bug in the jabber module. where do i open ticket for bugs ? |
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| 04:44.11 | ChannelZ | saint_: https://issues.asterisk.org/jira |
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| 05:24.20 | saint_ | is digium.com down ? |
| 05:25.34 | saint_ | or.. and so asterisk.org |
| 05:25.36 | saint_ | damn it |
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| 05:42.50 | zerohalo | saint_: don't seem down to me |
| 05:44.17 | ChannelZ | http://www.downforeveryoneorjustme.com/ |
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| 06:49.37 | din3sh | gd mrning all |
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| 08:22.17 | tat | has anyone experience with asterisk on osx, is it possibill to use my motorolla sm 56k modem receive and send calls using my landline ? |
| 08:22.31 | ectospasm | tat: no |
| 08:22.36 | ectospasm | a modem does not transmit voice |
| 08:23.43 | igcewieling | The few modems which are capable of "transmitting voice" are designed to handle voicemail and are not capable of realtime bidirectional voice. |
| 08:24.03 | igcewieling | Regardless, there are no asterisk drivers for them anyway. |
| 08:24.06 | ectospasm | tat: you'd need something like a Digium TDM or AEX410, with at least one FXO module... and I don't know the status of the DAHDI drivers on OSX |
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| 08:30.13 | tat | thanks, for the fast answer |
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| 08:32.11 | phpboy | Hi, I have a hosted PBX, I've got SIP connections from point A over the internet to the hosted PBX and point B. when I make calls the caller's channel is clear but the callee's channel is jittery |
| 08:32.27 | phpboy | what's the most common cause for something like this? |
| 08:33.12 | ectospasm | probably the network connection to point B, I'd guess. |
| 08:33.34 | ectospasm | do you know what kind of capacity the link has? |
| 08:33.41 | ectospasm | phpboy: ^ |
| 08:40.13 | ectospasm | if it's truly network jitter (packets arriving out of order), you might try the jitter buffer settings |
| 08:48.27 | phpboy | well, the connection seems stables from point A to point B |
| 08:48.54 | phpboy | getting under 50ms response with no packet loss |
| 08:49.06 | phpboy | I thought it may be the link between the two points but it's not |
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| 09:01.26 | ChannelZ | What is the bandwidth of both links? |
| 09:05.51 | phpboy | point A is 4Mbit ADSL, point B is 100Mbit hosted |
| 09:07.30 | phpboy | So I'm pretty confident bandwidth isn't the issue here |
| 09:07.40 | phpboy | well hopefully :T |
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| 09:26.31 | danfromuk | Hi. Is there anything within asterisk that can send a warning email if an outgoing calls results in a '500 service unavailable'? |
| 09:28.45 | kaldemar | danfromuk: your options are to examine the DIALSTATUS variable and then do what you want. |
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| 09:40.48 | danfromuk | Is there an active OP in the channel? |
| 10:03.13 | phpboy | lol, seems it was a school boy problem... codec conversion issue |
| 10:03.18 | phpboy | problem fixed :D |
| 10:03.42 | ectospasm | phpboy: that'll getcha every time |
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| 10:12.41 | phpboy | ectospasm: out of interest sake, there's no way around that issue right? |
| 10:13.05 | phpboy | would love to do G 729 on the one end and G 711 on the other |
| 10:20.09 | ectospasm | no reason it can't be done... probably need to be mindful of the packetization. |
| 10:20.29 | ectospasm | e.g. if it's 20ms per packet on one end, prolly should have 20ms on the other |
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| 10:20.47 | ectospasm | see the packetization text file in the Asterisk source/doc directory |
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| 11:04.24 | BorjaGVO | Hi, first of all, I know that this might not be the correct channel, but I don't see where I can ask for this. I'm trying to receive 2+ calls in a softphone but I don't find any that supports it. I see this very basic feature. Can anyone recommend a softphone or tell me if I must enable any options in Asterisk? Thank you very much. |
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| 11:27.28 | BorjaGVO | I know that activating call waiting (*71), several incoming calls from other users can be reveived. However, I'm trying to receive several calls on one same softphone within a queue but it's not working? Is it possible? |
| 11:36.28 | EmleyMoor | Why am I getting the error shown on line 210 of http://paste.debian.net/231373/ when I try to send a fax using t38modem? Is there anything I can do about it? |
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| 12:02.04 | EmleyMoor | I added t38pt_rtp = yes and now get http://paste.debian.net/231905/ |
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| 12:04.59 | gavimobile | why did my google calendar stop connecting.res_calendar_caldav.c:157 caldav_request: Unknown response to CalDAV calendar myGoogleCal, request REPORT to /calendar/dav/me@gmail.com/events/: Could not read status line: connection was closed by server |
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| 12:12.56 | EmleyMoor | Is there any way to do faxes simply (with or without hylafax) on Asterisk 1.8? I've got a way to send, it's just receiving that's a pain. |
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| 12:21.45 | bipul | Hellow Any one have any idea about Network Simulation with emulation ? does any one know how to do Voip emulation on ns-2. |
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| 13:36.25 | bipul | Hellow Any one have any idea about Network Simulation with emulation ? does any one know how to do Voip emulation on ns-2. |
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| 13:46.48 | WIMPy | Simulation with emulation? What does that even mena? |
| 13:50.23 | bipul | WIMPy, search for NS-2 emulation. |
| 13:50.55 | bipul | WIMPy, That is completely call "nse" |
| 13:51.20 | WIMPy | That doesn't mean anything to me. |
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| 14:05.42 | leifmadsen | ATDG 4e is now done. Thanks to everyone for playing along! |
| 14:08.01 | Kobaz | wow |
| 14:08.05 | Kobaz | my vision is pretty good today |
| 14:08.43 | WIMPy | remember he wantes to take a look... |
| 14:13.02 | *** join/#asterisk Rico29 (~rico@oceanet-telecom-fttb-129-2.olm.fr) |
| 14:13.06 | Rico29 | hi all |
| 14:13.25 | Rico29 | I'm having trouble while trying to register a trunk from my asterisk box to an opensips server |
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| 14:15.19 | Rico29 | it works well with other equipments (like oneaccess, etc...) |
| 14:15.25 | Rico29 | but not with asterisk |
| 14:15.42 | leifmadsen | waits for the punch line |
| 14:16.14 | jmetro | on the contrary, life without oxygen is a danger. |
| 14:16.21 | kaldemar | Rico29: what kind of trouble? |
| 14:17.29 | Rico29 | kaldemar> opensips always returns "wrong password" |
| 14:17.34 | Rico29 | but the password is the good one |
| 14:18.11 | kaldemar | where do you see that "wrong password"? what do you see in sip debug? |
| 14:18.44 | kaldemar | what does your register statement look like? |
| 14:20.05 | Rico29 | register statement : xxx.fr:5060 Y 1001 300 No Authentication |
| 14:20.27 | Rico29 | and in cli : WARNING[23444]: chan_sip.c:20458 handle_response_register: Forbidden - wrong password on authentication for REGISTER for '1001' to 'xxx.fr' |
| 14:20.59 | [TK]D-Fender | Rico29, pastebin the actual registration attempt |
| 14:21.17 | Rico29 | the sip debug ? |
| 14:21.18 | *** join/#asterisk shadar (~eugene@37.113.202.81) |
| 14:22.16 | kaldemar | Rico29: by register statement i mean the line in your sip.conf. and yes, pastebin the whole sip debug for a registration attempt. |
| 14:22.40 | [TK]D-Fender | Rico29, Clearly |
| 14:23.07 | [TK]D-Fender | Rico29, Any special characters in your password? Three are several that * does not support |
| 14:23.18 | Rico29 | non special chars |
| 14:28.10 | [TK]D-Fender | Rico29, Checked your realm? |
| 14:28.32 | Rico29 | [TK]D-Fender> yes |
| 14:28.47 | jmetro | werent you going to post configs we can check? |
| 14:28.56 | [TK]D-Fender | Rico29, So pure alphanumeric PW? |
| 14:29.01 | Rico29 | yes |
| 14:29.16 | Rico29 | it's in the pastebin I've sent you |
| 14:29.28 | [TK]D-Fender | So that's "actual"? |
| 14:29.29 | [TK]D-Fender | hrm |
| 14:29.31 | Rico29 | (super secure passwd) |
| 14:29.32 | Rico29 | yes |
| 14:30.35 | [TK]D-Fender | It's all so basic.... well give the OpenSIPS side a few more go-overs but it all feels pretty normal here... |
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| 14:38.27 | Rico29 | [TK]D-Fender> looks like the nonce calculated by asterisk is wrong... |
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| 14:48.12 | EmleyMoor | Is there anywhere I could generate a fax to myself from, other than myfax.com? |
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| 14:55.53 | rue_house | ups store? |
| 14:56.15 | rue_house | copy centres usually have machines you can use |
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| 15:04.13 | Rico29 | [TK]D-Fender> looks like this problem : http://lists.digium.com/pipermail/asterisk-users/2012-January/269012.html |
| 15:06.07 | [TK]D-Fender | Rico29, What version are you running? |
| 15:08.19 | Rico29 | 1.8.9 |
| 15:08.21 | Rico29 | .1 |
| 15:08.31 | Rico29 | I'm compiling latest 1.8 |
| 15:08.41 | Rico29 | but why mysql modules are marked as deprecated ? |
| 15:08.50 | Rico29 | will mysql support be disabled ? |
| 15:09.29 | *** join/#asterisk deo (~deo@112.198.82.44) |
| 15:09.31 | Greenlight | I think the idea is to use mysql via odbc moving forward |
| 15:09.58 | *** join/#asterisk psykon (~joel@mail.vicimarketing.com) |
| 15:10.19 | Rico29 | ok, is there any big changes to do for replacing mysql by odbc ? |
| 15:11.28 | *** join/#asterisk nanoha-sama (~nanoha-sa@2001:470:e97f:1003:215:5dff:fe07:4806) |
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| 15:12.34 | Rico29 | Greenlight> ? |
| 15:12.37 | Greenlight | I can't remember setting up mysql via odbc being too tricky |
| 15:13.01 | Greenlight | But, the old mysql module should still work if you want to use that |
| 15:13.23 | Greenlight | Although it's marked as deprecated, I seem to recall you can still include it |
| 15:14.12 | Rico29 | ok |
| 15:16.13 | *** join/#asterisk Tarcert (c1a9b80e@gateway/web/freenode/ip.193.169.184.14) |
| 15:19.35 | Rico29 | [TK]D-Fender> do you think it can be my problem ? |
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| 15:20.28 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:20.53 | [TK]D-Fender | Rico29, these are the sort of things that would be more commonly found bugs and possibly fixed. No reason to be behind on this so far.... |
| 15:21.19 | [TK]D-Fender | Rico29, And still a good idea to check the changelog/tracker to see if something specific was mentioned. |
| 15:21.39 | Rico29 | yep |
| 15:21.41 | Rico29 | ok |
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| 15:29.17 | Katty | g'morning lads. |
| 15:33.59 | aberrios | nope |
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| 15:50.09 | leifmadsen | Katty: OH GOD! MY ASTERISK IS ON FIRE! |
| 15:50.16 | leifmadsen | I'm losing dataz by the minutes! |
| 15:51.38 | chris_n | hands leifmadsen a bucket of water and another cup of coffee |
| 15:51.55 | leifmadsen | drinks the coffee before doing anything else |
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| 15:57.45 | Katty | coffee is a very important part of a complete morning. |
| 16:00.33 | Katty | oh what should i start today? |
| 16:00.42 | Katty | i watched the Once Upon a Time series...and finished Firefly up |
| 16:00.45 | Katty | what next? |
| 16:01.10 | Katty | something good to pre-occupy my brain between tickets. |
| 16:01.13 | _Corey_ | Katty: I just started watching Homeland... it's pretty good |
| 16:01.18 | Katty | what's it about? |
| 16:01.23 | sruffell | I take it you already watched Battlestar Galactica? |
| 16:01.45 | Katty | sruffell: mmmm, not all of it |
| 16:01.54 | sruffell | ok..you know what you need to do... |
| 16:01.56 | Katty | sruffell: i got up to the point where they discovered 13 flags on a moon, i believe |
| 16:02.00 | _Corey_ | Katty: it's a spy thing... there's a "Manchurian Candidate" quality to it |
| 16:02.14 | Katty | russell is right tho |
| 16:02.23 | Katty | i should finish up battlestar galactica. it's epic. |
| 16:02.34 | Katty | _Corey_: but that sounds like a good one to watch too! |
| 16:08.22 | *** join/#asterisk dwayne (~dwayne@c-71-207-208-112.hsd1.al.comcast.net) |
| 16:13.53 | Weezey | anyone know if it's possible to change an exension button on a 7960/7940 to show status of a queue or something else? |
| 16:14.41 | [TK]D-Fender | Weezey, XML browser on it. |
| 16:16.27 | *** join/#asterisk navaismo (~navaismo@189.191.239.96) |
| 16:16.28 | Weezey | [TK]D-Fender, but the user would have to press a button to see that, I'm thinking like change an icon, flash an extension. |
| 16:16.50 | Weezey | I guess I could use a mailbox and flash new voice mail, but that would be quite confusing. |
| 16:16.55 | Weezey | for idiots. |
| 16:17.00 | Weezey | like my users. |
| 16:17.35 | Weezey | [TK]D-Fender: know of any way via the telnet to change the name of an extension button? |
| 16:17.47 | igcewieling | Weezey: Cisco seems to reserve all the cool features for Skinny/SCCP and CallManager |
| 16:18.18 | Qwell | Weezey: Digium phones can. </unsolicited_advertisement> |
| 16:18.38 | igcewieling | As can Polycom. I bet Linksys SPAs could as well. |
| 16:22.24 | anonymouz666 | igcewieling: and do you know how reliable is SKINNY/SCCP support in Asterisk? |
| 16:22.50 | [TK]D-Fender | Weezey, Does thier XML include an "idle" screen? Polycom & Aastra do... |
| 16:30.41 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 16:30.41 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.2.1 (2013/01/22), 10.12.1 (2013/01/22), 1.8.20.1 (2013/01/22), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 16:30.46 | Qwell | Katty: the ship itself is alive. |
| 16:30.49 | apb1963__ | Yes... they were all aliens |
| 16:30.53 | Katty | ah. then this isn't lexx i'm thinking of |
| 16:31.00 | apb1963__ | Probably not |
| 16:31.04 | apb1963__ | Nobody was blue |
| 16:31.05 | Katty | the alien interfaced with the ship |
| 16:31.10 | Katty | similiar to the borg, but not the borg. |
| 16:31.14 | apb1963__ | no |
| 16:31.23 | apb1963__ | Stanley was the ship's captain |
| 16:31.46 | apb1963__ | Ki was the dead assassin |
| 16:31.48 | aberrios | you're thinking of Farscape Katty ? |
| 16:31.49 | Katty | this is going to drive me absolutely insane |
| 16:32.02 | apb1963__ | Farscape had a blue alien |
| 16:32.04 | Katty | YES,. that one |
| 16:32.12 | apb1963__ | She was a plant |
| 16:32.21 | Katty | shows you how much i remember about the series |
| 16:32.30 | Katty | some crazy alien at the center of a ship, and some guy with a weird alien beard. |
| 16:32.38 | apb1963__ | Yes... with multiple arms |
| 16:32.46 | apb1963__ | Yes.. that was farscape |
| 16:32.52 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:32.54 | aberrios | Farscape was the one where everyone dies at least once and comes back to life |
| 16:33.00 | apb1963__ | He could knock people out with his tongue (the beard guy) |
| 16:33.15 | apb1963__ | That one was with Claudia Black |
| 16:33.29 | Katty | i don't remember anything about lexx tho. |
| 16:33.33 | Katty | other than the girl looks familiar. |
| 16:33.38 | Katty | battlestar galactica, then lexx. |
| 16:33.45 | apb1963__ | then Supernatural |
| 16:33.50 | apb1963__ | then Fringe. |
| 16:33.50 | Katty | i've seen supernatural |
| 16:33.54 | apb1963__ | ok |
| 16:34.01 | Katty | i've seen parts of fringe i believe |
| 16:34.08 | apb1963__ | you have to watch it from the beginning |
| 16:34.20 | Katty | or maybe i'm thinking of Monk |
| 16:34.25 | apb1963__ | season one, episode one |
| 16:34.34 | Katty | either way, i lean more towards the scifi than the Crime Stuffs |
| 16:34.34 | apb1963__ | Monk was the defective detective |
| 16:34.52 | Katty | tho Bones was pretty good. |
| 16:34.55 | apb1963__ | Fringe had two universes |
| 16:34.57 | Katty | Lost Girl was decent, tho a little cheesy |
| 16:35.17 | Katty | i got completely Lost in Lost...i missed a few episodes and suddenly nothing made sense with alternate realities and what not |
| 16:35.24 | apb1963__ | Walter the Scientist... and Walternate |
| 16:35.30 | Katty | i'm going to have to start over with Lost |
| 16:35.45 | Katty | which series is shorter....Battlestar Galactica or Lexx? |
| 16:35.50 | apb1963__ | Walter Bishop |
| 16:36.04 | apb1963__ | Hmmmm.... not sure |
| 16:36.42 | apb1963__ | BG has movies and remakes so... possibly Lexx |
| 16:36.42 | Katty | there's 4 seasons of lex. |
| 16:37.09 | Katty | 4 seasons of battlestar galactica...but a lot more episodes per season. |
| 16:37.12 | Katty | so maybe i'll start with Lexx |
| 16:37.23 | apb1963__ | BG starts with Lorne Greene in what... 1978? |
| 16:37.37 | Katty | i'm going to watch the newer BS series |
| 16:37.42 | Katty | BSG |
| 16:37.57 | apb1963__ | They're both good |
| 16:38.00 | Katty | 2003 |
| 16:38.12 | Katty | i've seen the old ones before. they're very good |
| 16:38.23 | Katty | never finished up the newer ones, so i'll tackle those after Lexx. |
| 16:38.25 | apb1963__ | I never finished either... couldn't find all the episodes |
| 16:38.41 | Katty | i'm just watching what's on netflix. |
| 16:38.43 | Katty | there's probably more. |
| 16:38.49 | Katty | but i can always buy the series if i enjoy them |
| 16:39.02 | jmetro | watch the original doctor who - season 12 with Tom Baker. The best. |
| 16:39.14 | Katty | i've seen a lot of doctor who, jmetro |
| 16:39.18 | Katty | i started on the very very very very first one. |
| 16:39.21 | Katty | with the cave men |
| 16:39.28 | apb1963__ | Unrelated... there's some web series that's amatuer and low budget but very funny and very well done IMHO.... D&D style. |
| 16:39.31 | Katty | and watched ALL of them up to all but this last season |
| 16:39.48 | apb1963__ | Yes I used to have about 40 seasons of Dr. Who |
| 16:39.58 | Katty | the first ones were hard to get into |
| 16:40.03 | apb1963__ | watched most of them |
| 16:40.19 | Katty | the old black and white ones. cave men....then the daleks. |
| 16:40.27 | Katty | those first few episodes. i felt like i might fall asleep ;) |
| 16:40.36 | Katty | tv was a lot different back then. |
| 16:40.42 | apb1963__ | The first and maybe second episode were very good... the rest... not so much for many, many seasons. |
| 16:40.59 | apb1963__ | Agree |
| 16:41.08 | Katty | anywho. i'mma watch lexx now...so mostly afk (= |
| 16:41.17 | apb1963__ | waves. Enjoy |
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| 17:13.35 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-vrvgdpacevppgxcx) |
| 17:16.20 | curfont | urgent.. if I want to forward a number a call, the header is set to the incoming call that I am forwarding, how do I change that header? |
| 17:16.59 | curfont | For example.. what do I put before "Dial" to make it my account number for the voip provider |
| 17:17.13 | WIMPy | Set the caller ID |
| 17:17.26 | WIMPy | That's a different thing. |
| 17:18.09 | curfont | They said this" You should 1. Adjust the from header, instead of the mobile number it should be your account number." |
| 17:18.16 | curfont | The mobile number being the forwarded one |
| 17:18.28 | curfont | So I should change the caller Id to my account number? |
| 17:18.50 | WIMPy | Looks like your peer is incomplete. |
| 17:19.03 | curfont | Set(CALLERID(name)=Asterisk PBX) ? |
| 17:19.04 | curfont | This? |
| 17:20.07 | curfont | still didn't work, hmm |
| 17:20.16 | curfont | Is the header she is saying different to the callerid? |
| 17:20.48 | curfont | "from header" |
| 17:20.59 | igcewieling | curfont: generally CALLERID(num) is what sets the "user" in the From header |
| 17:21.29 | igcewieling | CallerID(name) is both totally cosmetic and totally useless when talking to most providers, it should not cause an issue though. |
| 17:23.10 | *** join/#asterisk Hive (~Hive@173-165-205-1-jacksonville.hfc.comcastbusiness.net) |
| 17:23.26 | curfont | igcewieling: " Conflicting extension values given." |
| 17:23.33 | curfont | Asterisk is spitting that out now |
| 17:23.39 | curfont | But it's using the correct "from" header |
| 17:24.08 | igcewieling | curfont: where are you getting that error message? |
| 17:24.50 | igcewieling | if in the CLI pastebin the CLI output leading up to and including that message. pastebin.ca |
| 17:25.13 | curfont | [Feb 5 17:24:51] NOTICE[1652]: chan_sip.c:27435 sip_request_call: Conflicting extension values given. Using X and not Y |
| 17:25.16 | curfont | thats all |
| 17:25.19 | curfont | After the dial |
| 17:25.27 | curfont | it's a NOTICE |
| 17:25.37 | igcewieling | I eagerly await the CLI output leading up to that error |
| 17:26.08 | Hive | Hello Asterisk people, is there a maximum value for priority inside of a context? I tried setting a priority to a 10 digit phone number, then doing a GoTo to that priority, but Asterisk does a GoTo to 1866989779 (notice the 9 digits :[ ). |
| 17:26.32 | igcewieling | Hive: 1) likely 2) don't do that |
| 17:26.38 | curfont | igcewieling: Answer, Set, Dial, with my username and password in it.. |
| 17:26.40 | curfont | Thats all |
| 17:26.48 | curfont | Then that notice |
| 17:27.02 | WIMPy | Hive: Interesting experiment, but probably not really what you wanted. |
| 17:27.05 | igcewieling | curfont: Either pastebin the ACTUAL output or find someone else to help you. |
| 17:27.31 | WIMPy | curfont: It's about the peer in your sip.conf. |
| 17:27.34 | igcewieling | only mask the password |
| 17:28.07 | Hive | igcewieling: I'm trying to route calls depending on what number they were called in on. I currently have a bunch of "GoToIf" and then compare a number to the number called in on(please dont scold me too much). This works fine for tiny amounts of phone numbers, but larger customers are spamming my CLI with literally hundreds of GoToIf statements. Any thoughts on how I should approach this challenge? :x |
| 17:28.46 | curfont | igcewieling: http://pastebin.ca/2310947 ? |
| 17:29.09 | WIMPy | Hive: Usually calls arrive in extensions, which are the dialled number. How exactely are they not doing that for you? |
| 17:30.27 | curfont | WIMPy: The incoming or dial out peer? |
| 17:30.41 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
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| 17:31.02 | WIMPy | curfont: The oue you're sending that call to. |
| 17:31.08 | Hive | WIMPy: well I created a "check-blacklist" context in which calls pass in their caller ID as the ${EXTEN}. Here we encounter some GoToIf's to see if the ${EXTEN} is in the blacklist. If we get through the blacklist, i then route the call based on the number called in on (i call it ${DNIS}). |
| 17:31.43 | igcewieling | curfont: I cannot help you further. |
| 17:32.03 | curfont | igcewieling: Why? I pasted the output.. |
| 17:32.18 | WIMPy | Hive: 1. Why do you use EXTEN for the caller ID? 2. Goto an extension, not a priority and 3. I'm not sure all channel support dnid. |
| 17:32.28 | igcewieling | Hive: your best bet is to use exten => lines to match each DID |
| 17:32.51 | gavimobile | my google calendar I cannot get to reconnect. it was working and I don't know how to get it to work again! WARNING[4522]: res_calendar_caldav.c:157 caldav_request: Unknown response to CalDAV calendar motekpc, request REPORT to /calendar/dav/myact@gmail.com/user/: Could not read status line: connection was closed by server |
| 17:33.15 | gavimobile | im on day 5 trying to get it to work |
| 17:33.26 | curfont | WIMPy: I figured out what it means, it's trying to use my account number (CALLERID(num)) as an extension |
| 17:33.32 | curfont | For some reason |
| 17:33.46 | curfont | Instead of the number I put for the extension.. |
| 17:34.33 | Hive | WIMPy: 1) ummmmm I'm going to use inexperience at the time of creating this as my excuse here ^_^' 2) I think I should probably separate my blacklist out and my call routing into separate contexts. Then I can use ${EXTEN} as it seemingly is intended to be used. 3) I actually set ${DNIS} to a cahnnel variable when the DID is called, so i can guarantee it is set for ever call. |
| 17:35.19 | WIMPy | Yes, seperate contexts are definitely the way to go. |
| 17:35.22 | curfont | WIMPy: OK fixed it :) |
| 17:35.46 | Hive | Thanks for the input WIMPy and curfont :) |
| 17:35.49 | Hive | off i go! |
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| 17:50.05 | *** join/#asterisk kuku (~kuku@173-167-188-106-Illinois.hfc.comcastbusiness.net) |
| 17:50.13 | kuku | Any ideas why I would be getting this: X-Asterisk-HangupCause: Protocol error, unspecified |
| 17:53.52 | *** join/#asterisk protoboardmx (~protoboar@201.120.24.10) |
| 17:54.05 | protoboardmx | hello everybody |
| 17:54.49 | protoboardmx | I don't know if this is a good place to ask this but I need some help integrating our Avaya 1608 desk phones with Asterisk |
| 17:55.02 | protoboardmx | has someone done that before? |
| 17:56.05 | navaismo | the phones support SIP? |
| 17:56.09 | dpilon | H323 how fun |
| 17:59.29 | [TK]D-Fender | protoboardmx, Have you configured a peer for them in h323.conf ? |
| 18:01.55 | WIMPy | kuku: The "unspecified" part is the important one. |
| 18:02.40 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 18:03.19 | protoboardmx | I haven't configured anything yet |
| 18:03.29 | [TK]D-Fender | protoboardmx, Go try |
| 18:04.09 | protoboardmx | I know the phones use h323 and I also know that SIP can be enabled by upgrading the firmware of the phone |
| 18:04.14 | protoboardmx | I haven't done that either yet |
| 18:04.26 | protoboardmx | first I want to have more information |
| 18:05.01 | [TK]D-Fender | protoboardmx, ask away.... |
| 18:05.15 | citywok | what specific information are you looking for, other htan "how do i do it" ? |
| 18:06.08 | protoboardmx | ok, sorry for asking such an ambiguos question |
| 18:06.37 | protoboardmx | I meant ambiguous |
| 18:07.04 | citywok | you can do s/ambiguos/ambiguous/ |
| 18:07.16 | citywok | s/you/youuu/ |
| 18:07.29 | WIMPy | You don;t even have to use whole words. |
| 18:07.34 | igcewieling | Ambigui OS |
| 18:07.50 | citywok | s/u/y |
| 18:07.53 | citywok | s/u/y/ |
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| 18:16.22 | *** join/#asterisk vedic (~V@183.82.82.82) |
| 18:17.34 | vedic | Hello friends, I am trying to install Dahdi on my system which is connected with phone lines. make and make install worked fine but make config is giving errors: |
| 18:17.39 | vedic | /sbin/chkconfig --add dahdi |
| 18:17.39 | vedic | /sbin/insserv: No such file or directory |
| 18:17.39 | vedic | dahdi 0:off 1:off 2:off 3:off 4:off 5:off 6:off |
| 18:17.39 | vedic | make[1]: *** [config] Error 1 |
| 18:18.33 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 18:18.36 | WIMPy | What is insserv? |
| 18:20.14 | WIMPy | If your distro doesn't have it, you probably want to ignore that. |
| 18:20.44 | gavimobile | just posted on asterisk forum.. if anyone has expirence with res_calendar.so please respond http://forums.asterisk.org/viewtopic.php?f=1&t=85623&p=183332#p183332 |
| 18:22.58 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200:221:6aff:feb8:e0b2) |
| 18:23.09 | *** join/#asterisk DarthExpeditor (~DarthExpe@rrcs-71-43-76-226.se.biz.rr.com) |
| 18:23.16 | navaismo | .i dont know but i think this "connection was closed by server" says a lot |
| 18:23.58 | gavimobile | navaismo: what gave it away? |
| 18:25.00 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.92.60) |
| 18:25.45 | vedic | WIMPy: so can I assume that make config went successful? It shows that Error 1 |
| 18:26.19 | WIMPy | I don;t know what the script does. Do you have a config? |
| 18:26.49 | vedic | WIMPy: I just did make config after successful make install |
| 18:27.31 | WIMPy | Wait. Make config only instlls the init scrip, does it? |
| 18:28.12 | Qwell | vedic: Did you Google the error? That is a bug in your distro. |
| 18:29.08 | vedic | Qwell: I am using ubuntu 10.04 64bit. Do you really think its a bug? I did search on Google but didn't find that its a bug. Could u point me to that? |
| 18:29.29 | WIMPy | Why is that a bug? |
| 18:30.15 | WIMPy | And do you need a script anyway? |
| 18:30.15 | dpilon | vedic: google...it is hard --> http://loginroot.com/ubuntu-12-04-64bit-sbininsserv-no-such-file-or-directory/ |
| 18:30.39 | Qwell | https://www.google.com/?#q=chkconfig+%22insserv%3A+No+such+file+or+directory%22 |
| 18:31.55 | vedic | dpilon: thanks |
| 18:32.27 | vedic | WIMPy: init script is useful when server boots. Else dahdi will need to be run manuallyu |
| 18:32.51 | WIMPy | Shouldn't that be done by udev? |
| 18:35.14 | *** join/#asterisk CunningPike (~CunningPi@d28-23-24-84.dim.wideopenwest.com) |
| 18:38.37 | vedic | WIMPy: What is the procedure? Asterisk books says make, make install and make config |
| 18:39.23 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-199-109.red.bezeqint.net) |
| 18:40.03 | WIMPy | I have never tried to do make config. |
| 18:40.37 | WIMPy | I'm not a fan of tons of scripts anyway. |
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| 18:41.24 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.187) |
| 18:42.43 | ruben231 | hi guys i ahve installed asterisk 1.8 and i notice when i used asterisk -rvvvvvvvvvvvvvv and as i see the CLI logs its only white text no variation with color like, yellow,pink, - the usual asterisk have...any idea what could be wrong..? |
| 18:43.36 | WIMPy | Your teminal type? But IIRC the one when starting the daemon is important as well. |
| 18:48.16 | ruben231 | WIMPy:how do i check terminal type, i using ubuntu server |
| 18:48.24 | kuku | 'tcp' is not a valid transport type when tcpenable=no. If no other is specified, the defaults from general will be used << what am i doing wrong ? |
| 18:49.07 | *** join/#asterisk Devon_ (~Devon_@63.214.236.169) |
| 18:54.22 | ruben231 | WIMPy:..? you there |
| 18:55.01 | igcewieling | kuku: add tcpenable=yes in sip.conf [general] |
| 19:05.02 | *** join/#asterisk doctorray (~ray@72.26.99.19) |
| 19:07.03 | WIMPy | check $TERM |
| 19:07.07 | WIMPy | is off to the cinema |
| 19:11.22 | *** join/#asterisk TarCert (c1a9b80e@gateway/web/freenode/ip.193.169.184.14) |
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| 19:20.57 | ruben231 | WIMPy: how do i check it..? other than asterisk my directory and files the color hinting are fine and ok but tiwh asterisk CLI, its all white only |
| 19:21.24 | ChannelZ | echo $TERM |
| 19:23.26 | *** join/#asterisk navaismo (~navaismo@189.191.239.96) |
| 19:29.44 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.187) |
| 19:31.58 | jpsharp | ruben231: Did you start asterisk with the -c option? |
| 19:32.37 | jpsharp | oh, nevermind. Dumb idea. |
| 19:33.02 | *** join/#asterisk cmendes0101| (~cmendes01@wtnl.corp.tierra.net) |
| 19:33.08 | jpsharp | I thought that was the color option. |
| 19:34.51 | *** join/#asterisk mathi (~Matthew@91.179.214.209) |
| 19:34.53 | mathi | hi |
| 19:35.53 | mathi | I would like to put two persons in a call. How can I do that ? The idea is to ring one person, and when he takes the calls, ring the other person. Is this possible ? |
| 19:38.06 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 19:39.17 | jpsharp | mathi: Use the Asterisk originate command or call file. In the originate command, tell it to dial the second number as the Application/Data. |
| 19:41.11 | mathi | jpsharp, but wait ... I need to be able to do two outbound calls simultaneously right? |
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| 19:48.40 | mathi | leifmadsen, isn't that covered in your book by the way ? :) |
| 19:48.47 | leifmadsen | probaly |
| 19:48.52 | leifmadsen | didn't read back, but I'll say yes |
| 19:48.58 | leifmadsen | check the cookbook actually |
| 19:49.01 | mathi | uhm what chapter ? |
| 19:49.03 | mathi | ok |
| 19:49.15 | leifmadsen | otherwise, check the AMI chapter |
| 19:49.31 | leifmadsen | not sure.. there are like 29 chapters in the book :) |
| 19:56.30 | navaismo | ~book |
| 19:56.30 | 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 |
| 19:56.53 | navaismo | always lost the link |
| 20:04.33 | ruben231 | WIMPy: im using this ----------->xterm |
| 20:09.01 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.92.60) |
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| 20:27.41 | ruben231 | WIMPy: ..? still there |
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| 20:32.21 | ruben231 | hi guys i ahve installed asterisk 1.8 and i notice when i used asterisk -rvvvvvvvvvvvvvv and as i see the CLI logs its only white text no variation with color like, yellow,pink, - the usual asterisk have...any idea what could be wrong..? |
| 20:32.21 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 20:33.01 | jmetro | you can set the verbosity and debug level once and just connect with asterisk -r |
| 20:33.09 | jmetro | your terminal colors might be off. |
| 20:33.41 | ruben231 | <PROTECTED> |
| 20:34.36 | jmetro | no clue, google-fu |
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| 20:44.27 | navaismo | ruben231, look at asterisk.conf |
| 20:45.56 | *** join/#asterisk tamiel (~tamiel@208.66.27.62) |
| 20:47.54 | ruben231 | navaismo: then, what part..? |
| 20:48.48 | navaismo | ¬¬ |
| 20:49.01 | navaismo | scroll down check the options |
| 20:49.11 | *** join/#asterisk dpilon (~Any@c-50-138-178-238.hsd1.ct.comcast.net) |
| 20:50.52 | ruben231 | ;nocolor = yes ; Disable console colors. |
| 20:51.51 | ruben231 | thats what i see |
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| 21:33.39 | ketas | why is it that those dialplans are extremely difficult to write |
| 21:36.15 | igcewieling | ketas: because telephone is hard. pbxs are hard. |
| 21:36.26 | igcewieling | s/telephone/telephony |
| 21:40.45 | ketas | i remember i got something working in past |
| 21:43.43 | ketas | it's all sip at least |
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| 21:50.08 | ChannelZ | ruben231: is the main asterisk daemon has to have been run in an environment that supports color. |
| 21:52.55 | igcewieling | ChannelZ: Maybe he is starting it manually from rc.local or something rather than using safe_asterisk or the init script? |
| 21:53.50 | ChannelZ | Yeah usually it's that and either isn't being run with -c or whatever means the init script gets run by doesn't have a supporting terminal that * thinks it can do color on |
| 21:54.49 | ChannelZ | For whatever reason it seems like something changed for me where if I manually start asterisk with my init script from a shell running under 'screen', it doesn't work. |
| 21:54.52 | *** join/#asterisk nny (~Scott@mail.hhhealth.com) |
| 21:55.32 | ChannelZ | I feel like that started happening in * 11 as I don't remember it ever being like that before, though I can't really say for sure. |
| 21:56.00 | nny | quick question, I am setting up 2 servers, bridged via IAX2, with hints at one location. SIP is accessible to both. Currently they are on the same network, they will be moved to different networks eventually. How should the phones check for hints? Does the phone need sip accounts on both servers to do so? |
| 21:56.14 | nny | er with hints at both locations, sidecars at one location* |
| 21:56.28 | igcewieling | maybe caused by the console redirection stuff which was recently changed in the startup scripts? |
| 21:57.01 | ketas | hm, telephony hard... is it hard partly because it drags old bullshit along with it? |
| 21:57.52 | ChannelZ | igcewieling: maybe although I don't recall ever having updated my init scripts. |
| 21:58.35 | ChannelZ | I'm still using one I hacked up from 2008 :) |
| 21:59.19 | igcewieling | we start asterisk using either amportal (for FPBX installs) or the standard init script |
| 21:59.31 | jmetro | Ketas> Yes. |
| 22:00.15 | ChannelZ | the supplied debian one (I'm on Ubuntu) doesn't run safe_asterisk for LSB reasons so I'm using an old one that still does. Or rehacked it to. I forget. |
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| 22:01.44 | *** join/#asterisk nny2 (~Scott@cpe-174-107-223-014.sc.res.rr.com) |
| 22:01.49 | nny2 | sorry got dc'd |
| 22:02.03 | nny2 | if someone responded to my query could you repost? |
| 22:02.29 | ketas | damn, somehow i can find any proper examples too :( |
| 22:04.32 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:04.37 | igcewieling | ketas: what are you trying to do? |
| 22:05.18 | igcewieling | nny2: I don't believe hints are accessable between servers |
| 22:05.42 | nny2 | igcewieling: er I mean to say is the phone will try to register for hints at both locations |
| 22:05.52 | igcewieling | ask your phone? |
| 22:06.29 | nny2 | igcewieling: i was more interested in seeing how hints security works. The phone may not need to have two accounts on it to see both |
| 22:07.37 | igcewieling | I can't imagine the phones would be able to subscribe without an account in any sane world. |
| 22:09.22 | nny2 | igcewieling: is there a standard practice for hints and IAX2? |
| 22:09.37 | nny2 | i am kind of guessing at how it would work right now. I could be heading on the wrong path anyways |
| 22:10.19 | igcewieling | I was not aware hints were supported on IAX2, but I don't use it. IAX2 is more of an affectation, like wearing a fedora or bow tie or rolling your own cigarettes. |
| 22:13.10 | nny2 | oddly though suggested as the preferred method of bridging two systems over SIP by digium :\ |
| 22:13.11 | *** join/#asterisk igcewieling (~igcewieli@user-24-214-153-32.knology.net) |
| 22:13.18 | nny2 | oddly though suggested as the preferred method of bridging two systems over SIP by digium :\ |
| 22:13.19 | igcewieling | gcewieling: I was not aware hints were supported on IAX2, but I don't use it. IAX2 is more of an affectation, like wearing a fedora or bow tie or rolling your own cigarettes. |
| 22:13.29 | ketas | igcewieling: well, i have few sip phones and also few accounts from provider for in/out calling... i want to record all calls and also have voicemail answering if calls come in and no phone answers to them |
| 22:13.35 | igcewieling | when, in 2002? |
| 22:13.36 | ketas | igcewieling: not that difficult task?! |
| 22:14.04 | igcewieling | MixMonitor? |
| 22:15.25 | nny2 | igcewieling: maybe opinions vary, i'm not hellbent on it. I could just as easily use SIP, but that still leaves the core issue of hints |
| 22:17.38 | ketas | igcewieling: well, indeed |
| 22:23.21 | ketas | damn ffs |
| 22:23.34 | ketas | looks like impossible task |
| 22:23.34 | nny2 | ketas: i'd love to help you but stuck in my own issue atm |
| 22:23.52 | nny2 | ketas: but essentially you need a full dialplan it looks like |
| 22:23.58 | ketas | i know how to add sip user, i know how to add upstream... |
| 22:24.03 | ketas | doh, dialplans |
| 22:24.35 | nny2 | ketas: yeah you need to define your contexts, routes and rules for it. Have you done this before? |
| 22:24.44 | ketas | no |
| 22:24.53 | nny2 | ketas: do you have the asterisk book? |
| 22:24.55 | nny2 | ~book |
| 22:24.55 | 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 |
| 22:24.57 | ketas | no |
| 22:24.58 | *** join/#asterisk igcewieling (~igcewieli@235.sub-70-193-67.myvzw.com) |
| 22:25.00 | ketas | :P |
| 22:25.02 | nny2 | there it is for free :) |
| 22:25.28 | nny2 | it's not simple. It's kind of it's own language, you'd be best to start from the beginning. I hope you aren't in a hurry to set this up |
| 22:25.40 | ketas | no |
| 22:25.47 | nny2 | ketas: what's your goal right now and why? Just learning or single install? |
| 22:25.48 | igcewieling | Knology (ISP) has been SUPER reliable for years but in the past month I've had more service outages than in all of the previous 3 years |
| 22:25.51 | ketas | well i tried before |
| 22:26.14 | ketas | but never completed the dialing plan |
| 22:26.16 | ketas | :/ |
| 22:26.38 | ketas | nny2: learning |
| 22:26.51 | ketas | nny2: i would also want it to work after learning it |
| 22:26.56 | nny2 | ketas: i can link you my default one BUT I suggest using it to learn, just trying to implement it would be painstaking and you won't be able to do much with it |
| 22:27.57 | ketas | i remember that i got sip <-> sip local calls partly working, with voicemails |
| 22:28.02 | nny2 | ketas |
| 22:28.07 | nny2 | oops |
| 22:28.08 | nny2 | http://pastebin.com/uZqKc5W4 |
| 22:28.08 | hbee | if i would want to connect asterisk to a dx220 using ss7, would this be the cheapest way? http://www1.digium.com/en/products/telephony-cards/digital/single-span |
| 22:28.45 | nny2 | ketas: that is my base, mind you it's not what I end up with. some of it may even be outdated, but it shows the structure and purpose of contexts |
| 22:29.06 | nny2 | ketas: I'd try to learn what each of those contexts does, and implement your own. Like reading code backwards |
| 22:30.27 | nny2 | also looking at it there's a typo or two (old config) sip add heat should be sip add header. That line basically tells the phone to ring a certain way via a sip packet |
| 22:30.33 | igcewieling | "exten => s,n,SIPAddHeat-Info: n=Simple-1)" > |
| 22:30.40 | igcewieling | ??? |
| 22:30.54 | igcewieling | ah, you already saw it |
| 22:31.41 | nny2 | it warms the packet first |
| 22:31.49 | nny2 | no one likes a cold header |
| 22:32.28 | nny2 | looking at distributed hints, anyone have thoughts on it? |
| 22:32.51 | nny2 | looks like XMPP based sharing, wondering how it works in situ |
| 22:34.35 | ketas | phone in heat |
| 22:35.09 | ketas | nny2: now i need to simplify it out |
| 22:35.13 | ketas | damn |
| 22:36.04 | navaismo | im so bored :( |
| 22:36.53 | ketas | help me with dial plans :P |
| 22:38.56 | igcewieling | navaismo: take down a couple of customers. end of boredom problem |
| 22:39.05 | nny2 | ketas: you'll have to learn it. There's no easy way to explain it |
| 22:39.15 | *** join/#asterisk hbee (~hbee@87-100-155-185.bb.dnainternet.fi) |
| 22:39.28 | nny2 | navaismo: if you had 2 servers/locations via SIP how would you handle hints? |
| 22:39.56 | nny2 | i am looking at distributed hints right now |
| 22:39.57 | navaismo | igcewieling, no more customer :'( |
| 22:40.18 | navaismo | nny2, never used hints across servers, only local |
| 22:40.24 | ketas | nny2: i wonder why is it that i have no problem writing code, but i have great issues with dialplans? |
| 22:40.29 | nny2 | navaismo: gotcha, thanks |
| 22:40.42 | ketas | which should be basically same thing |
| 22:40.44 | nny2 | ketas: dialplans are more like BASIC imho |
| 22:40.52 | nny2 | ketas: do this, next, do this, goto here, etc |
| 22:40.59 | *** join/#asterisk KamiyK (~Thunderbi@l.oateshome.com.au) |
| 22:41.05 | igcewieling | nny2: AEL is a more modern looking enviroment |
| 22:41.31 | nny2 | ketas it's basically context | extension (exten => XXX,1 (step 1) APPLICATION (parameters) |
| 22:41.35 | nny2 | igcewieling: I agree |
| 22:42.42 | navaismo | what are the advantages using ael instead the normal dialplan? |
| 22:43.06 | *** join/#asterisk reconwireless (uid10170@gateway/web/irccloud.com/x-dwgpaackxldozmks) |
| 22:43.14 | nny2 | if you had [outbound dial] and exten => 1XXXXXXXXXX,1,Dial(${EXTEN}@SIPPROVIDER) and dialed 18435551212 (and your phone's sip context was outbound dial) it would then Dial(18435551212@SIPROVIDER) |
| 22:43.24 | jmetro | using scripts outside of dialplan, you can literally do anything except have feelings. |
| 22:43.30 | nny2 | hahaha |
| 22:44.05 | nny2 | sorry context should have been outbound_dial |
| 22:44.08 | nny2 | no space |
| 22:44.16 | ketas | jmetro: what if i need feelings? |
| 22:44.23 | jmetro | wait until the singularity. |
| 22:44.30 | nny2 | ok going to dig into this hint stuff. If there's a way to do it i'll figure it out |
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| 22:45.14 | nny2 | ketas: just read the book, you'll feel saner afterwards |
| 22:46.03 | nny2 | ketas: put it in your bathroom and eat a block of cheese |
| 22:46.24 | ketas | mh |
| 22:46.27 | ketas | paper book? |
| 22:46.34 | igcewieling | navaismo: The ONLY advantage to AEL over dialplan is AEL looks more like a language. The disadvantage is AEL is a lot harder to debug |
| 22:46.46 | nny2 | better imho unless you don't mind reading pdfs or converting it to an ebook |
| 22:47.25 | jmetro | The paper book helped me a lot when I started out, just making my own server on a Virtualbox on my laptop and getting a phone working with no outside help. |
| 22:48.08 | ketas | hmm |
| 22:48.42 | ketas | i have always read tech docs in non-paper form |
| 22:49.16 | ketas | you know, you can't search on paper |
| 22:49.35 | jmetro | paper isnt case sensitive either. |
| 22:49.46 | nny2 | http://www.voip-info.org/wiki/view/Asterisk+Documentation+1.6.1+distributed_devstate.txt is what I am looking at btw |
| 22:49.57 | nny2 | seems sane so far, need to set up a test and see how it works in practice |
| 22:50.16 | nny2 | ketas yeah just go with the free download then |
| 22:50.18 | nny2 | ~book |
| 22:50.19 | 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 |
| 22:50.33 | ketas | plus, you can't test things out right away |
| 22:50.40 | ketas | :P |
| 22:58.20 | *** part/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
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| 23:13.50 | ketas | maybe it's because i've never written basic |
| 23:14.00 | ketas | although i do understand it a bit |
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| 23:18.39 | *** join/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com) |
| 23:18.47 | nny | sorry may have been a bad analogy |
| 23:18.55 | nny | i meant it in terms of simplicity in structure |
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| 23:30.00 | ketas | and it doesn't help at all that all default config files are filled with things that you actually don't need |
| 23:31.03 | *** join/#asterisk solitude88 (~solitude8@wsip-24-234-107-109.lv.lv.cox.net) |
| 23:31.25 | nny | ketas move them to sip.conf.example, extensions.conf.example etc. I do this for voicemail sip and extensions.conf. the rest you'll touch on a need to basis based on the system |
| 23:31.31 | nny | ketas and create fresh ones |
| 23:33.51 | nny | can anyone tell me or link me the current suggested implementation for distributed hints? I see some different docs including this one https://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State+with+XMPP+PubSub |
| 23:34.44 | nny | leifmadsen: if you're around this is something you seem to have had a hand in at some point. Thanks in advance |
| 23:35.09 | leifmadsen | asteriskdocs.org |
| 23:35.10 | nny | from what I can gather XMPP and pubsub is better than openais |
| 23:35.22 | leifmadsen | s/better/different |
| 23:35.27 | leifmadsen | depends how your network topology is setup |
| 23:35.33 | leifmadsen | openais / corosync is much easier to setup |
| 23:35.38 | leifmadsen | but it can't be used outside of a lan |
| 23:35.40 | nny | leifmadsen: currently LAN, will be across WAN later |
| 23:35.46 | leifmadsen | then you need to use xmpp |
| 23:36.02 | leifmadsen | so you need to install and configure tigase in addition to asterisk |
| 23:36.11 | nny | may be a VPN between the two networks at some point though. I'll review both, thanks for your input |
| 23:38.43 | nny | leifmadsen: does latency affect one more than the other? I saw something mentioned about it but not specific |
| 23:39.03 | leifmadsen | latency affects corosync more because it can't be off the LAN |
| 23:39.08 | *** join/#asterisk DarthExpeditor (~DarthExpe@rrcs-71-43-76-226.se.biz.rr.com) |
| 23:39.19 | leifmadsen | it's not just a network topology thing; it's a latency thing |
| 23:39.31 | leifmadsen | it expects the network to have near-zero latency |
| 23:39.32 | nny | leifmadsen: roger, thanks |
| 23:39.42 | leifmadsen | VPN isn't likely to work very well |
| 23:40.19 | nny | leifmadsen: FYI I am looking into this for multi server hints. If there's a saner alternative I couldn't find any specifics. Otherwise it will be pubsup and XMPP |
| 23:40.36 | leifmadsen | whatever I know is documented in the book |
| 23:40.51 | nny | leifmadsen: gotcha, ok sorry to bother, thanks, perusing now, my copy is outdated |
| 23:41.08 | leifmadsen | ofps.oreilly.com has the 4th edition up for the time being |
| 23:42.00 | nny | leifmadsen: thanks again |
| 23:46.45 | *** join/#asterisk jsjc (~Adium@164.Red-2-136-102.dynamicIP.rima-tde.net) |
| 23:47.29 | ketas | ooh, i got something working again |
| 23:47.36 | ketas | jumps in happyness |
| 23:48.31 | igcewieling | I hate moving. 8-( |
| 23:48.43 | ketas | uh |
| 23:48.56 | ketas | igcewieling likes calmness |
| 23:49.22 | [TK]D-Fender | I hate being stationary |
| 23:49.28 | igcewieling | deciding what to keep and what to throw away. |
| 23:49.35 | [TK]D-Fender | Especially when all those idiots try writing on you.... |
| 23:49.43 | igcewieling | [TK]D-Fender: I usually move every 5-10 years. |
| 23:50.03 | igcewieling | [TK]D-Fender: and the pigeons! The pigeons! |
| 23:50.30 | igcewieling | I just need to keep reminding myself "Palm trees, palm trees!" |
| 23:51.31 | igcewieling | I'll cry when I throw away my Catalyst 5505 |
| 23:52.08 | ketas | i never move... |
| 23:52.15 | ketas | i'm happy here |
| 23:52.49 | ketas | oh, it works now |
| 23:52.54 | ketas | damn codec issues |
| 23:53.09 | igcewieling | thankfully I have a lot of really cheap stuff I got after Katrina, most of that will be thrown out. |
| 23:53.22 | igcewieling | ketas: best practice. disallow=all and only allow= the few codecs you want. |
| 23:54.50 | ketas | oh, i feel much better now |
| 23:58.53 | ketas | indeed there was codec issue |
| 23:59.02 | ketas | all was quiet :/ |
| 23:59.07 | ketas | for most of tries |