| 00:01.09 | linuxh | bmoraca_work, do you think that even not missing interrupts the timing shown by the dahdih_test being below the "you won't have problems" can be the cause?, the machine has less than .30 of load and more than 70 cpu power unused. |
| 00:02.52 | bmoraca_work | linuxh: more than likely, yes. that said, there's little you can do about it. this is why i don't advocate junk cards. invest in a decent 4-port Sangoma or Digium card with HWEC. it's your best bet. |
| 00:03.15 | linuxh | i use those tdm400, but this is for a friend.. no commercial value |
| 00:03.22 | riddlebox | or you can use oslec |
| 00:03.38 | linuxh | he can't afford those... (here anything is expensive) |
| 00:03.58 | linuxh | when usually there is no echo cancelation when entire tdm path |
| 00:04.07 | linuxh | when = well.. |
| 00:04.59 | linuxh | i see zaptel had an util that could display missed interrupts |
| 00:14.56 | linuxh | oh well, before bugging you guys any more.. i will lend to him a decent machine.. see if it fixes the problem |
| 00:15.00 | linuxh | thanx folks |
| 00:15.05 | linuxh | best regards from Brazil. |
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| 00:44.27 | magic_hat | howdy, folks. Our sip provider limits our account to 10 simultaneous channels. Is there any way I can have * check to see how many channels are in use before placing an outbound call? |
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| 00:46.08 | dlynes_laptop | magic_hat, GROUP_COUNT() |
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| 00:51.20 | magic_hat | dlynes_laptop: Cool. So if I'm reading this correctly, I can do something like exten => s,1,Set(GROUP=foo). and then GROUP_COUNT(foo) to make sure I don't overload our account? |
| 00:53.50 | voxter | anyone do any video conferencing in windows using sip/asterisk? |
| 00:54.05 | voxter | I know x-lite and such do 1-1 video conferencing, but im looking for a group video solution. |
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| 01:15.37 | ChannelZ | circle jerk? |
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| 01:29.52 | Sedorox | I'm having a interesting issue... using queues.. and addqueuemember/removequeuemember... the issue I have is if asterisk is restarted, the members stay in the db, and when they goto login, they are told they are already there, but it never rings them back (marked invalid)... so they have to logout, then log back in.. and I am runing 1.6.2.1.. anyone see this before? |
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| 01:32.38 | voipmonk | do u want to log them out or do u want asterisk to not remember when you restart |
| 01:32.49 | voipmonk | " i want the damn phone to ring..." |
| 01:32.51 | voipmonk | :) |
| 01:32.55 | Sedorox | lol |
| 01:32.58 | voipmonk | are you there yet? :) |
| 01:33.04 | Sedorox | basically wipe the db/logged out when it's restarted |
| 01:33.36 | Sedorox | as you never know what hapend between when it died.. and when it came back up.. and X person might not be available anymore... excluding that it doesn't even ring any of the extensions |
| 01:33.54 | voipmonk | do you use safe_asterisk ? |
| 01:34.01 | voipmonk | u can have it do that |
| 01:34.17 | Sedorox | hmm |
| 01:34.20 | voipmonk | sort of a hack |
| 01:34.25 | voipmonk | until u get the code fixed |
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| 01:35.11 | Sedorox | I'm using gentoo start/stop scripts... and which code would need fixed? |
| 01:35.51 | voipmonk | whatever u are using to spawn a new asterisk if the old dies |
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| 01:36.39 | Sedorox | gentoo init scripts... lol |
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| 01:56.31 | Docteh | hey if I do a Read() to a variable and then goto() an invalid extension will it drop to the i extension? |
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| 02:09.14 | rizwank | I'd like to log the length of calls into my asterisk box and record what # they call. (They're using Skype, so it's a Dial command) as well as track the incoming ANI of the caller and how the call ended. I could do all of this, except that if the person calling into my box hangs up, the dialplan terminates immediately. Is there a way of handling this? |
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| 02:10.26 | rizwank | oh, wait, is that a perfect use for the h extension? |
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| 02:19.19 | ChannelZ | rizwank: could be. There is also CDR that * already does |
| 02:19.41 | magic_hat | Howdy, folks. I'm messing about with the GROUP_COUNT function to make sure that an app I have doesn't exceed the number of channels available for outbound calls. Does this make sense? |
| 02:19.44 | magic_hat | http://pastie.org/794514 |
| 02:19.46 | rizwank | it will track if there are multiple Dials in a sessioin? |
| 02:20.45 | magic_hat | also: anyone know whether the call will still show up in the GROUP_COUNT if it's been handed off to the agi application? |
| 02:20.46 | ChannelZ | what do you mean, like Dial(SIP/someone&SIP/someoneelse) ? |
| 02:31.06 | rizwank | ChannelZ : like if you hit * to dial again. |
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| 02:50.02 | Katty | okay you geeks who like cars. |
| 02:50.19 | Katty | do you think you could power a small bird bath heater, around 50 watts,off a car battery? |
| 02:51.40 | dlynes_laptop | Katty, what does a battery have to do with liking cars? |
| 02:52.09 | dlynes_laptop | Katty, how many amps is the battery? |
| 02:52.18 | Katty | idk |
| 02:53.04 | florz | well, any car battery will provide some 4 A ... |
| 02:55.35 | dlynes_laptop | Katty, the short answer is yes, the long answer is it depends on how many amps your car battery is will determine how long it can power the heater for |
| 02:55.54 | Katty | i bet there's a formula for that somewhere |
| 02:56.18 | dlynes_laptop | Katty, w = i*e, where 'w' is your watts, 'i' is your amps, and 'e' is your volts |
| 02:56.28 | florz | no, it doesn't depend on the battery's maximum current, but on its capacity |
| 02:56.30 | p3nguin | Most car batteries are not going to be less than 600 A. |
| 02:56.33 | Katty | yeah i know the pie formula. |
| 02:56.41 | Katty | but that doesn't give me time |
| 02:57.48 | florz | you have to divide the capacity (in ampere-hours, Ah) by the current (Amperes, A) |
| 02:59.17 | Katty | looks at a car battery |
| 02:59.49 | Katty | ooh car battery selection wizard. |
| 02:59.56 | p3nguin | $35 at Walmart |
| 03:00.08 | Katty | :< they have nothing about support bird bath feeders :P |
| 03:00.16 | p3nguin | hahahaha |
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| 03:01.34 | Katty | hmmmmmmmmm |
| 03:01.43 | Katty | car batteries are 12 volts? |
| 03:01.49 | p3nguin | yes |
| 03:02.03 | Katty | these bird bath heaters are designed to run on your typical 110 wall outlet |
| 03:02.09 | p3nguin | Unless you get a specialty battery, then it's 16V. |
| 03:02.11 | dlynes_laptop | Katty, well...not quite |
| 03:02.19 | dlynes_laptop | Katty, they're supposed to be 12v |
| 03:02.29 | *** part/#asterisk korihor (n=korihor@190.205.251.97) |
| 03:02.45 | dlynes_laptop | Katty, but when they're fully charged, they're 13.5 to 15v, depending on the type of battery and the manufacturer |
| 03:02.45 | p3nguin | If you want to run it off a car battery, you'll need to get an invertor to go with it. |
| 03:03.05 | Katty | ah okay. i got one of those |
| 03:03.09 | dlynes_laptop | Katty, and it's DC |
| 03:03.13 | Katty | yeah |
| 03:03.56 | p3nguin | Is there any reason you wouldn't just plug it into line voltage, say via extension cord? |
| 03:03.58 | Katty | how many amps is your typical car battery? |
| 03:04.07 | florz | just put 8 in series and everything should be fine =:-) |
| 03:04.09 | Katty | p3nguin: there's no external power outlets anywhere at the house |
| 03:04.09 | p3nguin | 600-1000 Amps |
| 03:04.36 | Katty | okay, let's use 1000 amps for easy math then |
| 03:05.11 | florz | yeah, you can operate some 250 of those heaters from a single battery |
| 03:05.55 | p3nguin | If you intend to charge it, then run it dead, then charge it, then run it dead, etc... consider buying a deep cycle battery instead of a starting battery. |
| 03:06.00 | Katty | does the amperage change when i goes through an inverter? |
| 03:06.04 | Katty | s/i/it/ |
| 03:06.27 | p3nguin | Yes, the Amperage is the available current at 12 VDC. |
| 03:06.55 | Katty | yeah but the inverter is gonna change it to 120ac |
| 03:06.58 | florz | the thing is called "current", and yeah, conservation of energy applies, and power is current times voltage |
| 03:06.59 | p3nguin | The invertor will be where you need to concentrate. |
| 03:07.19 | p3nguin | You can get 100W - 1500W invertors. |
| 03:07.23 | Katty | does that mean it's 120v ac @ 1000amps? |
| 03:07.27 | Katty | oh ise e |
| 03:07.28 | p3nguin | nope |
| 03:07.31 | Katty | k, so that depends on the inverter |
| 03:08.08 | magic_hat | Folks, I'm messing about with the GROUP_COUNT function to make sure that an app I have doesn't exceed the number of channels available for outbound calls. Does this make sense? http://pastie.org/794514 . Also, anyone know whether the call will still show up in the GROUP_COUNT if it's been handed off to the agi application? |
| 03:08.16 | p3nguin | There are probably some really piddly invertors of even less Wattage. |
| 03:08.34 | florz | but I don't quite understand what the point of this exercise is - do you seriously want to move the battery inside once a day for recharging? |
| 03:08.52 | Katty | well i don't have my inverter in front of me |
| 03:09.27 | p3nguin | Is it moderately large in size and has large cables, or is it kinda smallish with a cigarette lighter plugin? |
| 03:10.05 | Katty | it's a cigarette lighter plugin |
| 03:10.12 | Katty | which wouldn't really be... |
| 03:10.13 | Katty | useful |
| 03:10.29 | p3nguin | You have options as far as that goes. |
| 03:11.22 | p3nguin | You could cut the plug off, you could tap into the wires without cutting the plug off, or you could get an auxilliary power port, which resembles a cigarette lighter socket. |
| 03:11.46 | p3nguin | I would get the socket so I didn't have to alter my invertor. |
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| 03:12.33 | florz | you almost certainly don't want to leave a cigarette lighter connection loaded with 4 A unsupervised |
| 03:13.00 | p3nguin | http://www.bellautomotive.com/asp/product_details.asp?id=25215 |
| 03:13.23 | p3nguin | Why would you say that? |
| 03:13.33 | p3nguin | 4A @ 12V is nothing. |
| 03:13.54 | florz | 4 A is 4 A, independent of the voltage |
| 03:14.45 | florz | and those cigarette lighter thingies tend to have rather high contact impedances |
| 03:15.33 | florz | especially if left to corrode outside, I would suspect :-) |
| 03:15.42 | p3nguin | Don't be a pansy. I've ran 20 A through cigarette lighter sockets. |
| 03:16.14 | florz | and it got pretty hot, right? |
| 03:16.22 | p3nguin | warm, not hot. |
| 03:16.33 | p3nguin | 4 A is nothing. |
| 03:16.45 | florz | then you had one of the better versions and had it plugged in firmly :-) |
| 03:17.04 | p3nguin | I would have ran more through it, but the wiring for it wouldn't support it. |
| 03:17.10 | Katty | okay |
| 03:17.32 | Katty | so a typical car batter, providing 100 amp hours, would power my little 50 watt bird bath feeder for approximately 25 hours |
| 03:17.55 | p3nguin | So you would have to charge it every day. |
| 03:18.00 | Katty | if it was running constnatly, pulling 4.somethingish amps |
| 03:18.05 | p3nguin | And it probably takes 8 hours to charge. |
| 03:18.19 | Katty | :< |
| 03:18.36 | Katty | sighs |
| 03:18.58 | Katty | maybe i'll just stick a kerosene heater under the bird bath :P |
| 03:19.05 | p3nguin | Any reason you can't set up a solar charger? |
| 03:19.11 | Katty | well yes. |
| 03:19.13 | Katty | it's missouri |
| 03:19.20 | Katty | when do we ever have bright sunny middle of winter days? |
| 03:19.20 | p3nguin | oh |
| 03:19.40 | p3nguin | They're kinda rare, I suppose. |
| 03:19.43 | Katty | it's always cloudy and blahhh |
| 03:19.53 | Katty | i was just going to get a solar powered bird bath |
| 03:20.06 | Katty | until i realized the power generated from its solar panel in Direct Sunlight could only keep the water moving |
| 03:20.34 | Katty | it's not strong enough to keep the water from freezing, and it doesn't have any storage cells |
| 03:20.48 | p3nguin | bother. |
| 03:21.11 | p3nguin | I think I would just add exterior power and call it a day. |
| 03:21.18 | Katty | i don't know how to do that |
| 03:21.38 | Katty | katty is not qualified to play with electrical wiring yet :P |
| 03:21.45 | Katty | wiring up a light switch doesn't count |
| 03:21.59 | p3nguin | It's not too much different. |
| 03:22.17 | Katty | not gonna do it |
| 03:22.28 | Katty | and ryancan't be bothered |
| 03:22.48 | Katty | but not to fear, there's always hot water from the coffee machine to warm the birdy bath up (= |
| 03:29.26 | dlynes_laptop | Katty, yeah...and meanwhile your birds are going to be flying high as a kite from all the caffeine |
| 03:29.38 | Katty | ^_- |
| 03:29.42 | Katty | i seriously doubt that |
| 03:29.47 | dlynes_laptop | hehe |
| 03:29.51 | Katty | i don't drink coffee |
| 03:30.17 | dlynes_laptop | Katty, then why do you have a coffee machine? |
| 03:30.32 | Katty | ryan drinks coffee like once a month |
| 03:30.36 | Katty | but i drink hot tea |
| 03:30.46 | Katty | everyone has a coffee machine |
| 03:30.55 | Katty | everyone has a microwave >.< |
| 03:31.02 | dlynes_laptop | shurg |
| 03:31.06 | dlynes_laptop | i don't have a coffee machine |
| 03:31.11 | dlynes_laptop | i use a water boiler for my tea |
| 03:31.46 | Sedorox | I don't drink coffee |
| 03:31.54 | Katty | that would also work |
| 03:31.56 | p3nguin | I don't have coffee machine, and I don't drink coffee. |
| 03:31.58 | Katty | a kettle would be easy |
| 03:38.06 | Katty | ryan's cranky )= |
| 03:38.42 | Katty | apparently amerenue changed up their website. |
| 03:39.30 | p3nguin | Good to know they're doing something with all that extra money from the rate increase. |
| 03:42.11 | *** join/#asterisk AeroCloud (n=aero@ip72-222-149-220.ph.ph.cox.net) |
| 03:44.53 | AeroCloud | anyone had a SIP termination that sends 200 OK before the call is actually answered? if so do you know of a fix? |
| 03:49.45 | [TK]D-Fender | AeroCloud: Tons do and there is no fix |
| 03:50.05 | [TK]D-Fender | AeroCloud: They consider the call answered and billable the moment they hit the PSTN on their side |
| 03:54.32 | AeroCloud | I know.. |
| 03:54.51 | AeroCloud | I can track my side with voice detection, but I need a fix |
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| 03:58.07 | AeroCloud | I wonder why noone has developed voice detection for providers like this |
| 03:58.22 | AeroCloud | within the asterisk code.. not dialplan |
| 03:59.26 | AeroCloud | Another Question: Is it possible to send a Code after the call has been answered? |
| 03:59.32 | AeroCloud | from the dialplan |
| 03:59.48 | [TK]D-Fender | AeroCloud: Because progress is supposed to be OOB, not "audio" |
| 04:00.42 | AeroCloud | for example.. send a fake number.. like 2000 OK |
| 04:01.10 | AeroCloud | instead of 200 OK, but have the other asterisk server setup to accept 2000 instead of 200 |
| 04:01.19 | AeroCloud | to start the call |
| 04:03.03 | AeroCloud | I can do the patching for the 2000/200 thing.. not a problem.. its just sending the ack after a call is already in place |
| 04:03.10 | AeroCloud | from dialplan |
| 04:06.03 | voipmonk | what in the.... |
| 04:06.06 | voipmonk | u know what |
| 04:06.07 | voipmonk | nevermind |
| 04:06.11 | voipmonk | goes back under his rock |
| 04:07.41 | AeroCloud | whats up voipmonk |
| 04:09.06 | voipmonk | i have nothing good to say with what you are attempting so I wont say much more :) |
| 04:09.25 | AeroCloud | lol I have no choice its billing clients incorrectly |
| 04:09.28 | ManxPower-work | ARGH! I hate "web technologies". |
| 04:09.33 | AeroCloud | have to make something work |
| 04:09.40 | ManxPower-work | More like "web make it overly complicated" |
| 04:09.52 | voipmonk | AeroCloud: oh "it" is? :) |
| 04:10.04 | AeroCloud | yeah sip termination client |
| 04:10.05 | voipmonk | dont u hate "it" when "it" does that? |
| 04:10.06 | AeroCloud | yep |
| 04:10.11 | ManxPower-work | voipmonk: done any HTTP POST'ing of XML data? |
| 04:10.18 | AeroCloud | its to our benefit.. but still |
| 04:10.19 | voipmonk | so the sip termination client is the billing system? |
| 04:10.26 | AeroCloud | no |
| 04:10.31 | AeroCloud | they connect through us |
| 04:10.32 | voipmonk | lies, NOPE! |
| 04:10.41 | AeroCloud | and because its sending 200 OK before actually answered |
| 04:10.47 | AeroCloud | they are billing their clients wrong |
| 04:10.50 | voipmonk | jer might help, ManxPower-work |
| 04:11.06 | ManxPower-work | voipmonk: how does he feel about polycoms these days? |
| 04:11.27 | voipmonk | ok - i'll bite - |
| 04:11.33 | voipmonk | what are you trying to do , ManxPower-work |
| 04:13.06 | ManxPower-work | voipmonk: This: http://pastebin.ca/1766103 |
| 04:13.20 | ManxPower-work | I suspect I simply can't do it with libcurl in PHP. |
| 04:13.56 | ManxPower-work | The phone expects to have the request authenticated using Digest mode |
| 04:14.11 | voipmonk | thats sexy |
| 04:14.17 | ManxPower-work | And I can't seem to just send raw data using curl after the authentication. |
| 04:14.22 | ManxPower-work | voipmonk: It would be sexy if it worked. |
| 04:15.18 | voipmonk | OAI ? |
| 04:15.36 | ManxPower-work | OAI? |
| 04:20.59 | voipmonk | ManxPower-work: did you finish reading that polycom pdf? |
| 04:21.12 | voipmonk | web applications developers guide |
| 04:21.22 | voipmonk | sip 3.1 ( 77 pages ) |
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| 04:26.41 | AeroCloud | voipmonk, do you know a fix.. this could be a paid job |
| 04:27.10 | voipmonk | AeroCloud: are you playing middle? |
| 04:27.16 | AeroCloud | middle? |
| 04:27.21 | voipmonk | buying from 1 selling to another? |
| 04:27.22 | AeroCloud | yeah |
| 04:27.25 | AeroCloud | yes sir |
| 04:27.42 | AeroCloud | we buy large quantity's so its cheaper |
| 04:36.41 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 04:39.24 | frek818 | What do people do when the want Asterisk to be exposed to multiple SIP ports. |
| 04:39.40 | frek818 | bindport only allow you to specify a single port. |
| 04:44.11 | *** join/#asterisk [8none1] (n=[8none1]@cerberus.franklinamerican.com) |
| 04:45.47 | drmessano | Wow, sending UDP to a broadcast address results in packets going all over the effin network |
| 04:45.51 | drmessano | Who wudda thunk it? |
| 04:47.55 | ChannelZ | the guy who invented the broadcast address |
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| 05:03.57 | [TK]D-Fender | frek818: 0.0.0.0 <------- ................. --------......... (wait for it) .............. ALL |
| 05:06.10 | frek818 | Wait for what? |
| 05:06.31 | [TK]D-Fender | ... humour fail |
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| 05:14.11 | ChannelZ | also answer fail? he asked about ports, not interfaces |
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| 06:03.58 | docelmo | weee! |
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| 06:44.51 | AlexC | I'm trying to set up asterisk 1.6.1 as a SIP voip server for a group of people and can't an easy way. can someone please point me in the right direction? |
| 06:49.33 | p3nguin | alexc: AsteriskNOW |
| 06:50.32 | AlexC | i can't use that |
| 06:50.44 | p3nguin | What's the problem? |
| 06:50.49 | AlexC | i'm installing it on a server i don't have phisical access to |
| 06:51.18 | p3nguin | It already has an OS on it? |
| 06:51.24 | AlexC | yes |
| 06:51.28 | AlexC | ubuntu 9.04 |
| 06:51.32 | AlexC | 8.04 sorry |
| 06:51.34 | p3nguin | I guess it's time to start reading. |
| 06:51.39 | p3nguin | ~book |
| 06:51.40 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 06:51.57 | AlexC | there are no tutorials on the internet? |
| 06:52.05 | AlexC | oh |
| 06:52.16 | AlexC | my bad |
| 06:52.33 | p3nguin | ~wiki |
| 06:52.53 | p3nguin | ~wikis |
| 06:52.54 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 06:52.58 | AlexC | i've compiled i just need to configure |
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| 06:56.23 | AlexC | does the book cover 1.6? |
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| 07:22.52 | kruemeltee | hello all together ... :-) |
| 07:27.05 | TommyBotten | Good morning |
| 07:29.00 | enriquemora | Good morning to all |
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| 07:57.19 | enriquemora | I need to find a low-cost solution to connect a ISDN BRI to Asterisk. I need a TA (Gateway) or a quality card for less than 250Eur. Any suggestions? |
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| 08:30.48 | Polysics | hello |
| 08:30.59 | Polysics | ok, i am about to give up on the "flash web phone" thing |
| 08:31.03 | tzafrir_laptop | enriquemora, there are chepo single-port ISDN BRI cards supported by Asterisk |
| 08:31.17 | tzafrir_laptop | best supported ones are HFC-S -based ones |
| 08:31.18 | Polysics | too stupidly difficult to troubleshoot, and no real developer community to help |
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| 08:33.45 | Dovid | Polysics: If you really need it pay some one to do it for you |
| 08:34.39 | Polysics | Dovid, i actually was not able to locate ANYONE for that :-) |
| 08:34.48 | Polysics | i would be more than happy |
| 08:35.09 | Polysics | but apparently there is only a handful of red5 devs around |
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| 08:41.25 | enriquemora | tzafrir_laptop, thanks for the advice. I have seen some cheapo cards but they are Sooo cheap I am very skeptical |
| 08:43.36 | *** join/#asterisk af_ (n=getsmart@88-149-240-203.dynamic.ngi.it) |
| 08:43.55 | tzafrir_laptop | An adapter makes Asterisk to a (sort of) gateway |
| 08:44.10 | tzafrir_laptop | Anything that is mass-produced may be cheaper. |
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| 08:46.52 | Polysics | Dovid, being able to contact a developer on Google Code would be enough, but I can't fathom how |
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| 09:36.06 | shadey_ | Is it possible to have asterisk do a 'console dial' and automatically playback an audio file? |
| 09:36.15 | shadey_ | or something similar |
| 09:38.03 | creativx | yup |
| 09:40.06 | shadey_ | Playback(path/to/audio/file) ? |
| 09:40.07 | shadey_ | :-) |
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| 09:48.10 | benngard | yay!!! http://www.aastra.com/cps/rde/xchg/04/hs.xsl/15162.htm up and running :) |
| 09:48.43 | benngard | 1 step closer for me to throw our avaya to hell! |
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| 10:04.44 | Polysics | back to square 1 |
| 10:04.55 | Polysics | while i find someone i can pay for fixing this... |
| 10:05.33 | Polysics | how/can i be absolutely sure a client that is unable to transmit audio isn't a firewall/network issue? |
| 10:05.46 | Polysics | problem is, the client is actually running on the server |
| 10:06.00 | Polysics | i wonder if there is any kind of testing tool i could deply |
| 10:07.58 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:08.33 | TommyBotten | Polysics: Try using wireshark |
| 10:08.48 | Polysics | on the server? |
| 10:08.53 | Polysics | i'll go read up on wireshark |
| 10:08.56 | TommyBotten | Polysics: You can use tcpdump and log to a file, and view in wireshark later in |
| 10:10.41 | *** join/#asterisk DelphiWorld (n=Miranda@41.110.0.199) |
| 10:10.43 | DelphiWorld | hi |
| 10:10.50 | TommyBotten | Hello, DelphiWorld |
| 10:10.51 | DelphiWorld | anyone used Opensbc with asterisk? |
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| 10:20.12 | Polysics | from a summary tcpdump test, i would say the packets are there |
| 10:20.27 | Polysics | which further complicates issues |
| 10:21.29 | Polysics | it might be just about time to quit on this :-) |
| 10:21.35 | Polysics | activeX plug-in it is |
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| 10:23.59 | *** join/#asterisk _abc_ (n=no@unaffiliated/ccbbaa) |
| 10:24.02 | _abc_ | hello |
| 10:24.34 | _abc_ | has anyone seen a problem where a sip phone cannot call a unistim phone but the other way around works, and anything other than sip can call the unistim? |
| 10:24.53 | Polysics | i will have to make do with activex, people apparently can't live without a web based softphone |
| 10:25.07 | Polysics | i have been pointed to Zoiper, any other alternatives? |
| 10:25.22 | _abc_ | Polysics: noooo get php or somehting. there are java and flash web based phones already |
| 10:25.42 | Polysics | _abc_, they DO exist, none of them actually WORK :-) |
| 10:25.50 | _abc_ | Polysics: i disagree |
| 10:25.52 | _abc_ | sec |
| 10:25.55 | Polysics | and php is not related to them :-) |
| 10:26.11 | Polysics | well, if you can point me to a working flash phone, i'd be happy :-) |
| 10:26.47 | Polysics | red5phone is more of a PoC than a working implementation |
| 10:26.59 | Polysics | Zingaya Media Server is waaaaay overpriced, even if it DOES work |
| 10:27.11 | *** join/#asterisk _cgc (n=_cgc@94-193-99-128.zone7.bethere.co.uk) |
| 10:27.21 | _cgc | hi everyone |
| 10:27.57 | _cgc | does anyone know any good sip providers? |
| 10:28.43 | Polysics | _cgc, those are country-dependent :-) |
| 10:28.57 | Polysics | in Italy i am happy with messagenet, asterisk worked at first try |
| 10:28.59 | _cgc | Polysics: lol, ok then in the uk |
| 10:31.40 | Polysics | not in the UK, so can't help :-) |
| 10:31.53 | Polysics | _abc_, about those flash phones? if you know of a working one i'd love you :-) |
| 10:32.08 | _abc_ | Polysics: i said sec as in wait a little |
| 10:32.15 | Polysics | oh :-) |
| 10:32.52 | Chainsaw | _cgc: I'm in the UK, but we have ISDN BRI coming into the offices. |
| 10:34.17 | _cgc | Chainsaw: I have an isdn bri line as well but I want a sip trunk for backup, I got 1 with draytel but it's not working and their support aren't very helpful |
| 10:34.44 | Chainsaw | _cgc: *nod* |
| 10:35.46 | _cgc | Chainsaw: I'm looking at voip talk atm, had any expierence with them? |
| 10:36.29 | Chainsaw | _cgc: None, sorry. I've not used VoIP providers. I just use VoIP to do neat call routing with fallback paths internally. |
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| 10:44.30 | enriquemora | + |
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| 10:54.03 | Zhad | Has anyone here set up BLF on a snom 200? |
| 10:55.49 | Zhad | It seems no matter what format I enter the url in the Fn Keys dialogue, It doesn't subscribe for notifications. |
| 10:56.06 | Zhad | This is simple to set up on a GXP 20xx. |
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| 11:09.44 | E-bola | Hello all i got a pretty simple question thats confusing me. If i want to have a complete isolated context for a seperate branch of our company in asterisk. How do i handle incoming calls via our extern SIP->PSTN provider |
| 11:09.50 | E-bola | they normally come in via incoming |
| 11:09.59 | E-bola | is there some way to make them come in via the specified context? |
| 11:11.12 | E-bola | I had hoped it was enough to just specify a context on the account in sip.conf (which is used for calling out via the sip->pstn provider) doesnt seem to work |
| 11:12.28 | E-bola | hmm i stand corrected it seems to work |
| 11:12.39 | Zhad | afaik, that should be it, just change context= in the peer declaration. |
| 11:14.11 | E-bola | Yep, it apears to work, i coudl just swear i had tried it before where it didnt |
| 11:24.28 | Polysics | _abc_, got any news for me? :-) |
| 11:28.55 | _abc_ | Polysics: i am busy. there is a load of flash phones out there, none are really free, excepting the one from google (google talk) |
| 11:29.02 | _abc_ | keep looking of pay some money |
| 11:29.07 | _abc_ | and they all mostly work |
| 11:29.25 | Polysics | could you just say a few names? money is not a problem if they don't cost 1 million $ :-) |
| 11:33.45 | _abc_ | Polysics: flaphone ribbit tringphone "gizmocall flash phone" "google talk" |
| 11:33.53 | _abc_ | enough now, do your own homework |
| 11:34.43 | _abc_ | here i ask again: has anyone seen a problem where a sip phone cannot call a unistim phone using asterisk 1.6 ? where any other phone can call both, and callback etc works on both. |
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| 12:02.30 | Zhad | doesn't use unistim |
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| 12:11.16 | cadmium | hi, in my dialplan i call noOp(${EXTEN}) and it outputs the cli.. but when i use SayDigits(${EXTEN}) there is silence... SayDigits works if I just give it some numbers any-one understand what is happening ? |
| 12:11.48 | Chainsaw | cadmium: Perhaps you need double-quotes? |
| 12:14.05 | cadmium | Chainsaw nevermind i'm dumb... i had it commented the whole time i was trying different things |
| 12:14.24 | Chainsaw | cadmium: Ouch. Yeah, comments would do it. |
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| 12:52.40 | eth01 | can anybody SUGGEST an easy to get hold of motherboard for trixbox? |
| 12:53.00 | eth01 | doesn't have to be fancy, but needs to be reliable, stable and decent for a budget of £100 |
| 12:53.36 | TommyBotten | Most modern motherboards will do. When it comes to features, less is more in this case. |
| 12:53.47 | eth01 | ok so the ASUS p5e deluxe? |
| 12:53.50 | TommyBotten | eth01: You won't need HD sound, etc. |
| 12:54.08 | TommyBotten | I have good experiences with ASUS cards in general. Never tried the one you mention, though. |
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| 13:20.16 | kruemeltee | has anybody ever connected an elmeg telephone system on asterisk? |
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| 13:28.45 | [TK]D-Fender | kruemeltee: This is the low end of telephony choices for which I expect next to no-one is using here. |
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| 13:32.02 | kruemeltee | okay ... was just a question ;-) |
| 13:33.43 | [TK]D-Fender | kruemeltee: You stuck with one of these? |
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| 13:39.57 | coppice | what's so bad about elmeg. I thought their ISDN stuff was popular in europe |
| 13:43.27 | [TK]D-Fender | coppice: they have a small niche from what I've heard |
| 13:51.17 | Polysics | [TK]D-Fender, i have never asked you what you think of the project i'm doing and what would you do instead |
| 13:51.26 | Polysics | the whole web-based phone thing |
| 13:51.46 | Polysics | you are the local expert, so it makes sense for me to ask :-) |
| 13:52.08 | [TK]D-Fender | Polysics: Get those spineless wimps to install a proper soft-phone at a MINIMUM. I'd consider it a hiring test. If you can't follow instructions you don't deserve the job |
| 13:52.52 | [TK]D-Fender | Polysics: Expert? Not sure I'd say that. I'm just just sharp, with a lot of common sense and creativity. |
| 13:52.57 | Polysics | and how would you handle all the metadata about the communications? ie. call history, textual chat, document exchange while call is up |
| 13:53.09 | Polysics | video is a plus but not mandatory |
| 13:53.25 | Polysics | i am thinking the whole web-based thing IS bullshit :-) |
| 13:53.26 | [TK]D-Fender | Polysics: You have no back-end at all yet, right? |
| 13:53.45 | Polysics | not much to speak of, a few php scripts to put users in mysql |
| 13:53.52 | Polysics | le'ts say "no" for brevity |
| 13:55.01 | [TK]D-Fender | Polysics:this seems like VERY custom work. calls recorded with common "job # tags", etc. Same one used for archivng file transfers,e tc. |
| 13:55.24 | [TK]D-Fender | Polysics: this has a SharePoint kinda feel to it (SP = crap from everyone I've talked with). |
| 13:57.10 | Polysics | yeah, i am not even sure on how t ogo about it - web-based had some advantages, after all, like being easy to carry metadata |
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| 13:59.11 | [TK]D-Fender | Polysics: Big prject to say the least |
| 13:59.40 | Polysics | it is, and i could get some more people on it - just AFTER i figure out how to do it, i don't have to do it myself |
| 13:59.49 | *** join/#asterisk NicoleMun (n=ssharma@pool-173-63-185-74.nwrknj.fios.verizon.net) |
| 14:00.03 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-204-40-42.acanac.net) |
| 14:01.48 | Polysics | [TK]D-Fender, would an activeX/Java applet be on the same level as my Flash-based mess? |
| 14:02.04 | Polysics | i'd say they are slightly better, at least they are sort of proper clients |
| 14:02.23 | [TK]D-Fender | Polysics: No, its clearly 1 level less. But thing of the trouble you're asking for in attempting to accomodate incompetance |
| 14:02.44 | *** join/#asterisk Da-Geek (n=Da-Geek@62.189.17.99) |
| 14:04.02 | Polysics | it's not an incompetence problem, it's the same reason as why many people still have IE6 |
| 14:04.29 | Polysics | in the public/corporation world, they have old, closed setups they cannot update or essential software stops working |
| 14:07.20 | Dovid | morning Y'all |
| 14:07.37 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 14:08.53 | [TK]D-Fender | Polysics: Zoiper doesn't need to be "installed" and you can include the configs WITH it. Extract & execute. |
| 14:11.31 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:11.47 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:12.39 | kruemeltee | [TK]D-Fender, yes, I stuck with these ... I have to connect asterisk and this elmeg in some way ... first idea was t |
| 14:13.18 | voipmonk | What's an elmeg? |
| 14:13.29 | voipmonk | Is that like eggnog ? |
| 14:13.37 | kruemeltee | first idea was using asterisk as SIP provider, but I get no connection between elmeg and asterisk (certainly the elmeg registered on asterisk) |
| 14:13.45 | voipmonk | Nutmeg maybe? |
| 14:14.03 | voipmonk | Provider? |
| 14:14.20 | kruemeltee | elmeg is a telephone system with ISDN and optional with a VOIP-Module |
| 14:14.26 | [TK]D-Fender | kruemeltee: They seem to speak SIP.... have you seen it try to INVITE to *? |
| 14:14.44 | *** join/#asterisk chasing`Sol (n=rc4@41.196.104.127) |
| 14:14.49 | kruemeltee | yes, they speak SIP ... thats right ... |
| 14:15.11 | kruemeltee | I added a number within sip.conf and the elmeg system connects to * |
| 14:15.38 | kruemeltee | something like this appears at the console: |
| 14:15.41 | kruemeltee | -- Saved useragent "FEC SIP Proxy 4.0" for peer 9999 |
| 14:16.04 | [TK]D-Fender | kruemeltee: Go look at SIP debug and see if the ELMEG tries to send it a call |
| 14:16.09 | Katty | is not very happy with the quality of the new camera put up :< |
| 14:16.11 | kruemeltee | so this means the elmeg is connected to * |
| 14:16.44 | [TK]D-Fender | kruemeltee: means it registered. What it intends on DOING with this is another matter |
| 14:16.47 | [TK]D-Fender | Katty: Mew. |
| 14:16.51 | Katty | [TK]D-Fender: mew. |
| 14:16.54 | kruemeltee | thats one of my biggest problems ... if I try to call with a local connected SIP phone to the assigned number of elmeg I got an error 480 (currently busy, or something like that) |
| 14:17.00 | Katty | [TK]D-Fender: http://www.ustream.tv/channel-popup/the-nut-house-bird-bath |
| 14:17.18 | kruemeltee | if I try to dial from elmeg to the local connected telephones nothing appears within the console |
| 14:17.25 | Katty | [TK]D-Fender: it looks washed out...and blurry :< |
| 14:18.29 | dmast | g'morning all |
| 14:18.31 | kruemeltee | I'm currently dialing with two methods ... dial(SIP/9999) ... and dial(SIP/9999@IP_OF_ELMEG) |
| 14:18.39 | Katty | [TK]D-Fender: i imagine it will get worse as the day gets brighter. |
| 14:18.43 | kruemeltee | mornin' dmast |
| 14:19.39 | kruemeltee | currently I think it's a matter of elmeg's configuration ... but just at the moment ... :-) |
| 14:20.10 | [TK]D-Fender | kruemeltee: Did you enable SIP DEBUG at * CLI? |
| 14:20.12 | kruemeltee | and the manuals getting me biting within the table :.) |
| 14:20.17 | [TK]D-Fender | kruemeltee: I'm thinking not... |
| 14:20.21 | kruemeltee | 2 times ... |
| 14:20.37 | [TK]D-Fender | kruemeltee: Show me the Elmeg registering and the call attempt to * |
| 14:20.55 | kruemeltee | there was a lot of stuff ... but just if I'm dialing from the local telephones ... if I dial from elmeg nothing happens |
| 14:21.07 | kruemeltee | wait a second |
| 14:21.31 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-40-42.acanac.net) |
| 14:21.40 | Katty | hi monk |
| 14:21.43 | voipmonk | hi hi |
| 14:22.38 | Katty | my new camera is crap :< |
| 14:22.48 | Katty | not nearly as good as my other one. i just shan't do. |
| 14:22.57 | kruemeltee | [TK]D-Fender, http://kubuntu.pastebin.ca/1766551 |
| 14:25.01 | voipmonk | 229 and 240 - why would you receive that ? :) |
| 14:25.21 | voipmonk | 480 Tem Unavail |
| 14:26.48 | [TK]D-Fender | kruemeltee: -- Executing [9999@default:1] Dial("SIP/2001-083bb888", "SIP/22@192.168.22.244") in new stack <- you are sending this call unauthed |
| 14:27.07 | [TK]D-Fender | kruemeltee: As for the Elmeg's answer... not sure. Could be "busy". Could jsut be a config issue with it. |
| 14:28.17 | kruemeltee | unauthd? how do I authenticate? |
| 14:29.55 | [TK]D-Fender | kruemeltee: Dial using a PEER with username & password just like with any ITSP/phone/whatever |
| 14:30.11 | kruemeltee | okay |
| 14:30.12 | [TK]D-Fender | kruemeltee: Same way you did for your GS |
| 14:30.17 | kruemeltee | right |
| 14:31.34 | *** join/#asterisk benngard (n=benngard@213.88.138.230) |
| 14:32.11 | *** join/#asterisk tacubo (n=tacvbo@189.146.173.116) |
| 14:32.12 | *** join/#asterisk simplydrew (n=simplydr@198.7.249.227) |
| 14:32.17 | kruemeltee | wait a second ... I saw something here ... seems as if the call arrives at the elmeg ... |
| 14:32.47 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:32.52 | kruemeltee | the line 245 tells me that the elmeg routes the call right to the phone with ID 22 ... as I told elmeg withing its configuration |
| 14:34.35 | *** join/#asterisk rossand (n=aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 14:35.35 | [TK]D-Fender | kruemeltee: #162 is where YOU told it to dial 22 at that IP. |
| 14:35.48 | [TK]D-Fender | kruemeltee: So of course that's the "To:"" |
| 14:37.23 | kruemeltee | aha ... thats the answer ... |
| 14:39.22 | kruemeltee | hmm ... I'll try to get a complete log with different dial types ... which one do you think is the correct one? |
| 14:39.31 | kruemeltee | dial(SIP/9999) |
| 14:39.44 | kruemeltee | dial(SIP/9999@ELMEG_IP) |
| 14:40.05 | [TK]D-Fender | kruemeltee: What do valid #'s on the Elmeg look like? |
| 14:40.13 | kruemeltee | or dial(SIP/22@ELMEG_IP) ... where 22 is the internal MSN of the phone connected to the elmeg |
| 14:40.47 | kruemeltee | good question ... |
| 14:41.21 | kruemeltee | the elmeg is just configureable within a http-based Configuration tool |
| 14:41.34 | kruemeltee | (as I know) |
| 14:41.48 | [TK]D-Fender | kruemeltee: What would you dial on a PHONE inside of it? |
| 14:41.54 | kruemeltee | just 22 |
| 14:41.57 | kruemeltee | or 23 |
| 14:42.05 | kruemeltee | just the internal MSN |
| 14:42.12 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 14:42.29 | ManxPower-work | We don't care about "MSN"s. We care about SIP. What's on the other end does not matter |
| 14:42.57 | ManxPower-work | kruemeltee: you have an [ELMEG_IP] section in sip.conf? |
| 14:43.06 | kruemeltee | jep |
| 14:43.14 | ManxPower-work | good |
| 14:43.48 | kruemeltee | looks like that: http://kubuntu.pastebin.ca/1766581 |
| 14:44.03 | ManxPower-work | What is still not working? |
| 14:44.11 | *** join/#asterisk e-jones (n=jkastner@nat/redhat/x-zbdrjotcpstfamah) |
| 14:44.33 | ManxPower-work | kruemeltee: You lied. That is a [9999] section, not a [ELMEG_IP] section |
| 14:44.47 | kruemeltee | so I don't have an elmeg_IP section ... |
| 14:45.10 | kruemeltee | sorry, don't want to lie ;-) |
| 14:45.12 | ManxPower-work | So what is your issue? |
| 14:45.42 | kruemeltee | connecting an elmeg telephone system with * |
| 14:45.56 | ManxPower-work | So what is the problem? |
| 14:46.16 | ManxPower-work | Dial(SIP/remoteexten@sipconfentry) |
| 14:47.02 | kruemeltee | yea ... great idea ... @sipconfentry would be [ELMEG_IP] entry, right? How should this one have to look like? |
| 14:47.29 | ManxPower-work | kruemeltee: I don't know. What is the elmeg expecting? |
| 14:48.16 | ManxPower-work | Does it require authentication or does it auth based on IP address? What codecs does it want? What's it's IP. Set it up similar to how you have set up your other IP phones |
| 14:49.41 | kruemeltee | but I thought, I just have to use an entry like [9999] ... the elmeg log in into * as a SIP telephone, just like the others ... with its number, host=synamic and its passphrase ... |
| 14:49.55 | ManxPower-work | Do it |
| 14:50.09 | kruemeltee | wait a second |
| 14:53.17 | ManxPower-work | Dial(SIP/remoteexten@sipconfentry) <-- not just a good idea, it's how Asterisk works. |
| 14:53.26 | kruemeltee | right |
| 14:53.51 | kruemeltee | but elmeg isn't asterisk ... |
| 14:55.01 | ManxPower-work | That does not matter. You are configuring Asterisk. You are connecting to it via SIP. Asterisk does not care and does not know and will no understand what you have on the far end. |
| 14:56.46 | kruemeltee | right ... so if I'm configuring a number [9999] in sip.conf like I did and elmeg registers as 9999 with this parameters I think asterisk has to work with this ... |
| 14:56.58 | kruemeltee | and elmeg is connecting to asterisk ... |
| 14:57.01 | ManxPower-work | correct. |
| 14:57.07 | kruemeltee | -- Saved useragent "FEC SIP Proxy 4.0" for peer 9999 |
| 14:57.16 | ManxPower-work | Then you would Dial(SIP/29@9999) |
| 14:57.21 | kruemeltee | this means that elmeg correctly connected to asterisk, right? |
| 14:57.39 | kruemeltee | okay ... this case I've never used ... wait ... |
| 14:57.43 | ManxPower-work | "sip show peers" will tell you the IP of the registered device. |
| 14:58.48 | kruemeltee | wait ... I'll show you something |
| 15:00.33 | kruemeltee | http://kubuntu.pastebin.ca/1766596 |
| 15:02.15 | *** part/#asterisk benngard (n=benngard@213.88.138.230) |
| 15:03.04 | ManxPower-work | Your elmeg rejected the call. |
| 15:03.06 | [TK]D-Fender | krumGo read your Elmeg's docs or call them for supoprt |
| 15:03.28 | kruemeltee | okay ... so you think on asterisk's site evering's just okay, right? |
| 15:03.33 | [TK]D-Fender | ManxPower-work: Not a regular 'regection". Kinda of "dialplan prbably not configured right in it type of thing |
| 15:03.40 | [TK]D-Fender | kruemeltee: Possibly |
| 15:04.18 | kruemeltee | thanks ;-) |
| 15:04.50 | *** join/#asterisk af_ (n=getsmart@88-149-240-203.dynamic.ngi.it) |
| 15:05.37 | kruemeltee | thanks a lot for your help ... if I got something I let ya know :-) |
| 15:05.56 | *** join/#asterisk minotaur01 (n=minotaur@24.215.3.50) |
| 15:11.59 | *** join/#asterisk chilicuil (n=sistemas@unaffiliated/chilicuil) |
| 15:22.08 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 15:22.08 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:22.41 | *** join/#asterisk neurosys (n=neurosys@c-66-176-171-19.hsd1.fl.comcast.net) |
| 15:28.07 | *** join/#asterisk coppice (n=chatzill@234.157.17.210.dyn.pacific.net.hk) |
| 15:33.03 | _cgc | does anyone know of a good sip provider in the uk that works with asterisk? |
| 15:33.26 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 15:33.56 | *** join/#asterisk Heretic (n=fallen@dsl-246-124-132.telkomadsl.co.za) |
| 15:35.21 | [TK]D-Fender | ~itsplist-uk |
| 15:35.22 | infobot | i heard itsplist-uk is UK based ITSps include http://www.voiptalk.org/ http://www.voipon.co.uk/ http://www.gradwell.com/ and a few other tinpot companies you can dig up with google. |
| 15:35.47 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:36.22 | _cgc | thanks [TK]D-Fender, does anyone have any experience with voiptalk? |
| 15:37.25 | voipmonk | cant make or receive calls? pastebin.ca some debug, _cgc |
| 15:38.54 | _cgc | voipmonk: im looking for another sip provider, i would like to hear off someone who has had it working in * first though, I've got 1 with draytel but its rubbish and the support aren't that helpful |
| 15:39.48 | Zhad | uses voiptalk and gradwell |
| 15:54.05 | *** join/#asterisk Deeewayne (n=dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 15:54.05 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:58.10 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 16:00.46 | *** join/#asterisk ThoMe (i=tm@tm.muc.de) |
| 16:00.47 | ThoMe | hiho |
| 16:01.25 | Naikrovek | ..hiho |
| 16:01.43 | ThoMe | when i try to send an fax |
| 16:01.45 | ThoMe | i have this message: |
| 16:01.46 | *** join/#asterisk The_Boy_Wonder (n=vossel@asterisk/batman-developer/dvossel) |
| 16:01.48 | ThoMe | http://paste.keks.be/8690/txt |
| 16:01.51 | ThoMe | why? |
| 16:02.20 | Naikrovek | "Max retries exceeded to host 127.0.0.1" |
| 16:02.28 | Naikrovek | looks to me like the wrong host |
| 16:02.40 | Naikrovek | i don't know though |
| 16:02.46 | Naikrovek | hang out someone will help ya |
| 16:06.06 | ThoMe | Naikrovek: hm ok. thank you :-) |
| 16:09.46 | *** join/#asterisk magic_hat (n=geoffdou@207-229-176-225.grn-bsr1.chi-grn.il.cable.rcn.com) |
| 16:09.52 | magic_hat | Folks, I'm messing about with the GROUP_COUNT function to make sure that an app I have doesn't exceed the number of channels available for outbound calls. Does this make sense? http://pastie.org/794514 . Also, anyone know whether the call will still show up in the GROUP_COUNT if it's been handed off to the agi application? |
| 16:10.15 | bmoraca_work | ThoMe: your IAXmodem isn't properly registered. check its logs and see what it says |
| 16:10.43 | ThoMe | bmoraca_work: hmmm. ok |
| 16:11.13 | bmoraca_work | magic_hat: if the channel was assigned a group and is still alive, then it will show up in groupcount until that channel has been hung up |
| 16:12.25 | bmoraca_work | magic_hat: although your while loop at the top leaves a lot to be desired |
| 16:21.01 | magic_hat | bmoraca_work: lol. why's that? |
| 16:21.35 | bmoraca_work | Asterisk will use a lot of resources maintaining all of those channels while they're waiting for others to finish |
| 16:21.49 | ManxPower-work | TrendMicro thinks "pastie.org" is a "dangerous site". LOL! |
| 16:22.38 | bmoraca_work | you'd be better served having the external application that's telling asterisk to originate keeping track of when to send and when not to send more calls through |
| 16:23.22 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 16:23.36 | magic_hat | bmoraca_work: yeah, not a bad idea. I'm working with RAGI and it doesn't seem to have access to the GROUP_COUNT function, though. |
| 16:25.14 | bmoraca_work | magic_hat: i would use AMI to orginate the calls and have that application keep track of how many calls are active and when the channels are destroyed. that's just me, though. |
| 16:26.12 | magic_hat | cool. it's weird. The GROUP_COUNT never seems to get above 1 or 2. not sure whether that's because the program is bombing, or the calls are all to disconnected numbers. |
| 16:26.38 | Katty | thinks about swapping the cameras around |
| 16:26.54 | bmoraca_work | you'd have to monitor your console to determine what the case is. NOOP your hangupcause or some such. |
| 16:27.10 | bmoraca_work | magic_hat: i can assure you, though, that groupcount does work and it does go above 2 :) |
| 16:27.26 | magic_hat | lol. that would be pretty funny. |
| 16:28.03 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:28.18 | *** join/#asterisk casix (n=casix@xenpbxedifici.adamvozip.es) |
| 16:30.04 | casix | hello |
| 16:30.10 | *** join/#asterisk smooth_penguin (n=smoove@59.95.26.177) |
| 16:30.13 | Katty | considers going to best buy and picking up another webcam |
| 16:30.35 | eppigy | YOU LEFT ME FOR A CHEAP TOY |
| 16:30.50 | Katty | i still love you |
| 16:31.05 | Katty | eppigy: did you see the new stream? |
| 16:31.13 | Katty | eppigy: it's like...useless. |
| 16:31.24 | eppigy | I has not |
| 16:31.28 | eppigy | ilu2 |
| 16:31.28 | Katty | http://www.ustream.tv/channel/the-nut-house-bird-bath |
| 16:31.34 | Katty | look how useless that is |
| 16:31.48 | Katty | i don't even see the YARD, much less the bird bath |
| 16:31.56 | jro | Are there any tftp server alternative in linux aside from tftp-server/xinetd |
| 16:32.01 | eppigy | oh man |
| 16:32.02 | jro | *alternatives |
| 16:32.04 | eppigy | it is :< |
| 16:32.20 | eppigy | jro: im sure there are plenty |
| 16:32.20 | ManxPower-work | jro: why are you asking? (i.e. the might be a better way than tftp) |
| 16:32.27 | Katty | eppigy: but that's what i get for setting up an old POS webcam i had stuffed in the basement. |
| 16:32.29 | eppigy | you will probably have to compile it though |
| 16:32.47 | casix | I have a dialplan like: http://pastebin.com/m1d946e1b my problem is if I call to a 100@outbound-dept3 it goes to _XXX and not to 100 wich is included via [users-dept1]. How can I change this behaviour? |
| 16:32.55 | eppigy | Katty: I see :< |
| 16:32.58 | jro | I have a cisco ata 186 that wont pull from tftp-server even though a normal tftp client works fine |
| 16:33.00 | eppigy | you will do bettre next time |
| 16:33.04 | jro | was going to try something else |
| 16:33.20 | ManxPower-work | jro: Ah, OK. |
| 16:33.43 | Naikrovek | worked out a way to reboot polycom phones remotely in case anyone was wondering. |
| 16:33.47 | Katty | eppigy: the logitech quickcam pro9000 seems to be the most highly rated webcam on newegg. |
| 16:33.54 | Katty | Naikrovek: i'd be intersted. |
| 16:34.02 | ManxPower-work | Naikrovek: You followed the instructions on voip-info.org |
| 16:34.02 | Katty | Naikrovek: i assume other than pretending to make a change |
| 16:34.05 | Katty | Naikrovek: and hitting submit |
| 16:34.13 | Katty | Naikrovek: via the webby bit. |
| 16:34.19 | Naikrovek | ManxPower-work: no. |
| 16:34.25 | Naikrovek | Katty: yes other than that |
| 16:34.38 | ManxPower-work | Katty: Naikrovek is a jerk, not an idiot. 8-) |
| 16:34.49 | Naikrovek | ManxPower-work: i don't even click links to that wiki anymore, i just assume it's all out of date now |
| 16:35.09 | Naikrovek | thanks, ManxPower-work, for the 'not an idiot' part :) |
| 16:35.11 | ManxPower-work | Naikrovek: There are some things on the site that are useful. The Polycom info is one of those sections. |
| 16:35.13 | Katty | ManxPower-work: well now that wasn't very nice. |
| 16:35.14 | Naikrovek | love you too buddy |
| 16:35.16 | Naikrovek | :D |
| 16:35.16 | *** part/#asterisk jro (n=jaredo@ganondorf.loclhst.com) |
| 16:35.18 | Katty | ManxPower-work: fine example you are. |
| 16:35.30 | ManxPower-work | Katty: I never said that I wasn't a jerk. |
| 16:35.31 | Naikrovek | Katty: it's fine |
| 16:35.38 | Naikrovek | manx and i have an understanding finally |
| 16:35.40 | Naikrovek | i think... |
| 16:35.46 | Katty | retract claws. |
| 16:35.46 | Naikrovek | time will tell |
| 16:35.51 | Katty | retracts claws. |
| 16:36.06 | *** join/#asterisk btsteve (n=tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
| 16:36.28 | Katty | i wonder if i called best buy, if they could tell me whether or not they have this webcam in stock |
| 16:36.33 | Katty | are they smart enough to do that there? :/ |
| 16:36.33 | Naikrovek | well i'm using a powershell script instead of a perl script. |
| 16:36.47 | ManxPower-work | Naikrovek: no real understanding needed. If you are nice I'll help you. If you don't want my help, you tell me. |
| 16:36.48 | Naikrovek | Katty: usually |
| 16:37.01 | Katty | oh boy Home of the Geek squad on their IVR |
| 16:37.08 | ManxPower-work | Naikrovek: sipsak should also be able to be used to reboot the phones outside of Asterisk |
| 16:37.15 | Katty | i wonder if they wouldn't mind setting up the webcam on the streaming server |
| 16:37.21 | Naikrovek | never heard of sipsak |
| 16:37.53 | Katty | i wonder if the Geek Squad can handle linux. |
| 16:37.58 | Katty | it's probably Not Supported |
| 16:38.03 | Naikrovek | probably not |
| 16:38.21 | Katty | they'd probably recommend Windows 7 Home with IE 8 |
| 16:38.48 | Naikrovek | no |
| 16:38.50 | Katty | heh, the girl i'm talking to can't find the webcam |
| 16:38.58 | Naikrovek | they'll point you to the new computer section |
| 16:39.00 | voipmonk | they would crap themselves if they saw a hackintosh |
| 16:39.23 | Katty | heh, you're probably right |
| 16:39.46 | voipmonk | i installed osx on a bunch of bestbuy computers and left once... |
| 16:39.48 | Katty | but i'm not willing to give up my modded rhino centos box |
| 16:39.52 | voipmonk | that was fun |
| 16:40.03 | Katty | i mean, come on. do they have any computers with an LCD display on the front of it? |
| 16:40.05 | bmoraca_work | i really wish that coldfusion wasn't so damned verbose sometimes |
| 16:40.06 | Katty | which are RED |
| 16:40.10 | Naikrovek | voipmonk: it would be as if they saw a bicycle with a 10yo go 110mph. "this.. isn't... possible." |
| 16:40.16 | Katty | i think not |
| 16:40.23 | voipmonk | i should have taken pictures |
| 16:40.24 | Katty | my crittercam broadcasting server is Too Cool for them. |
| 16:40.36 | Naikrovek | voipmonk: really? I've never gotten a hackintosh going |
| 16:40.43 | Naikrovek | guess i need new hardware or something |
| 16:40.56 | Katty | eppigy: yay, best buy has webcam on sale for 85 |
| 16:41.03 | *** join/#asterisk helloritesh (n=hellorit@121.243.172.188) |
| 16:41.09 | Naikrovek | $85? seems high |
| 16:41.09 | Katty | eppigy: hopefully birdbath will be visiable after lunch :> |
| 16:41.12 | Naikrovek | for a webcam |
| 16:41.21 | Katty | it's because it's a 2MP webcam |
| 16:41.26 | Naikrovek | HD? |
| 16:41.28 | Naikrovek | ah |
| 16:41.29 | Katty | yes |
| 16:41.31 | voipmonk | is it a 720p hd webcam with great lowlight? |
| 16:41.33 | Naikrovek | that'll do it |
| 16:41.37 | Katty | voipmonk: ehhh, well |
| 16:41.40 | Katty | voipmonk: define Great |
| 16:41.41 | eppigy | Katty: sweet :D |
| 16:41.43 | Naikrovek | heh |
| 16:41.44 | Katty | voipmonk: i mean it /is/ a webcam |
| 16:41.48 | voipmonk | well |
| 16:41.57 | voipmonk | check liquidigital down unduh |
| 16:42.04 | Katty | voipmonk: i have a logitech c600 broadcasting the front yard. it does amazingly well with light fluctations |
| 16:42.05 | coppice | non of the webcams seem to go to 1920x1080 |
| 16:42.08 | [TK]D-Fender | Naikrovek: http://www.youtube.com/watch?v=nELqJwd0r0Y&feature=related |
| 16:42.08 | voipmonk | the agent v4 is what i bought off ebay for $50 |
| 16:42.09 | helloritesh | Folks, I want to record conversation for the first 1 minute of the bridged call. The monior app records the entire conversation which could be long and stress the CPU. Record applicaiton only records a single channel and mixing my create conversation alignment issues. Any suggestions? |
| 16:42.12 | voipmonk | now the agentv5 is out |
| 16:42.23 | Naikrovek | [TK]D-Fender: ooOOooh |
| 16:42.30 | kruemeltee | okay ... day is over ... see ya tomorrow :-) |
| 16:42.35 | Naikrovek | laters |
| 16:42.54 | Katty | coppice: this one's video capture resolution is up to 1600x1200 |
| 16:42.56 | Naikrovek | [TK]D-Fender: want |
| 16:43.01 | Katty | coppice: 24bit color |
| 16:43.02 | voipmonk | http://agent.liquiddigital.com.au/ |
| 16:43.31 | [TK]D-Fender | helloritesh: Originate a local channel to chanspy() the original channel and monitor it w/ an absolute timeout. |
| 16:43.50 | [TK]D-Fender | Naikrovek: "crotch-rocket" taken all too literally |
| 16:43.56 | coppice | Katty: there are a few that do 1600x1200, but not 1920x1080 |
| 16:43.59 | helloritesh | but that will record the entire call, isn't it? |
| 16:44.07 | Katty | coppice: well not video |
| 16:44.13 | Katty | coppice: they can probably take a photo that well |
| 16:44.15 | [TK]D-Fender | helloritesh: ABSOLUTe timeout |
| 16:44.23 | helloritesh | ah! |
| 16:44.36 | helloritesh | example? I will try and figure out otherwise.. |
| 16:44.49 | Katty | coppice: this one claims to do 8MP photos |
| 16:44.57 | Katty | coppice: not that i care about photos, i'm just doing live streaming. |
| 16:45.08 | voipmonk | the agentv5 does full 1920 x 1080 |
| 16:45.10 | Katty | coppice: what i like is logitech's ability to compensate for lighting |
| 16:45.33 | coppice | Katty: yeah, its odd those ones don't do 1920x1080. they have enough sensor pixels |
| 16:45.34 | Katty | coppice: it really does a fabulious job of keeping the color steady from 7am to 7pm |
| 16:46.20 | coppice | Katty: white balance is remarkably easy to achieve |
| 16:46.24 | *** join/#asterisk brezular (n=brezular@adsl-dyn34.91-127-11.t-com.sk) |
| 16:46.27 | Katty | nods |
| 16:46.30 | Katty | they have nice gain settings too |
| 16:46.58 | Katty | i tend to not play with it tho on manual settings else it'll get washed out in about an hour due to the light outside changing |
| 16:47.07 | gr0mit | anyone using chan_sccp? |
| 16:47.14 | Katty | the c600 was about mmm 60 bucks |
| 16:47.26 | bmoraca_work | Katty: need one with auto-iris then |
| 16:47.29 | Katty | so i'm hoping the pro 9000 at 90ish will be even better. |
| 16:47.48 | Katty | bmoraca_work: it does. |
| 16:47.51 | Katty | bmoraca_work: read up. |
| 16:48.12 | bmoraca_work | then you won't have to worry about manually changing light gain levels or white balance or anything like that |
| 16:48.20 | Katty | nope, i don't |
| 16:48.22 | *** join/#asterisk waa (n=waa@balrog.credipar.com.br) |
| 16:48.30 | Katty | it's on auto settings right now, and the front yard is beautiful. |
| 16:48.34 | *** join/#asterisk paulc (n=paulc@unaffiliated/paulc) |
| 16:49.18 | Katty | http://www.ustream.tv/channel/the-nut-house-bird-bath <- i'm upgrading /that/ webcam |
| 16:49.21 | Katty | which is about 4 years old |
| 16:49.24 | p3nguin | helloritesh: MixMonitor records both sides at the same time. |
| 16:49.55 | sbrath | I'm testing aastra and grandstream phones. Any pointers on BLF+asterisk 1.6.2+GSX2020 |
| 16:50.19 | Katty | http://www.ustream.tv/channel/squirrel-critter-cam <- this one is the c600 webcam. |
| 16:50.27 | Katty | Big Difference. |
| 16:50.31 | sbrath | The lights are all just green, I have the hints, and subscribe stuff setup |
| 16:51.03 | sbrath | and sip show subscriptions shows that the phone is "subscribed" |
| 16:51.25 | Katty | runs off to best buy |
| 16:51.53 | helloritesh | but mixmonitor doesn't allow specific length? Any suggestions. I am not able to figure out the Absolute timeout case |
| 16:53.59 | *** join/#asterisk diatonic (n=diatonic@mail.clearwater-research.com) |
| 16:54.20 | [TK]D-Fender | helloritesh: core show function TIMEOUT |
| 16:54.25 | helloritesh | thx |
| 16:55.08 | p3nguin | Isn't TIMEOUT() going to end the call? |
| 16:55.27 | helloritesh | yeah, the dynamic monitor extension will end |
| 16:55.29 | helloritesh | thats fine |
| 16:55.31 | helloritesh | the call will remain |
| 16:55.36 | p3nguin | okay |
| 16:55.39 | helloritesh | so using ChanSpy would be the way to go |
| 16:56.00 | helloritesh | just need to figure out how to create a dynamic channel (extension) with Chanspy and asbolute timeout |
| 16:56.21 | helloritesh | so a call comes-in and we will be forwarding it to another DID through Dial |
| 16:56.51 | *** join/#asterisk GameGamer43 (n=GameGame@69.129.142.83) |
| 16:56.55 | helloritesh | once bridged, I need to intercept using Chanspy (on the fly, for all calls, dynamically) |
| 16:56.55 | [TK]D-Fender | helloritesh: before you Dial, Originate your Chanspy call. |
| 16:57.12 | [TK]D-Fender | helloritesh: use the M() dial opt, etc, or whatever |
| 16:57.20 | helloritesh | ok |
| 16:57.24 | helloritesh | ah.. |
| 16:57.25 | helloritesh | got it |
| 16:58.19 | helloritesh | but that would bridge the incoming channel to Chanspy and if I timed-out chanspy, my incoming channel my go away |
| 16:58.39 | helloritesh | let me try it |
| 16:59.52 | *** join/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
| 17:00.19 | helloritesh | Got it.. Thanks a lot [TK]D-Fender |
| 17:00.22 | *** join/#asterisk slidesinger (n=slidesin@c-68-44-99-50.hsd1.nj.comcast.net) |
| 17:00.23 | helloritesh | appreciate it. |
| 17:00.29 | [TK]D-Fender | helloritesh: You're welecome |
| 17:01.33 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.121.163) |
| 17:02.43 | *** join/#asterisk Greek-Boy (n=Monching@41.188.154.137) |
| 17:03.31 | praet | anyone working with spa-3000? Im having trouble setting it up |
| 17:03.52 | p3nguin | Do you have a specific question about it? |
| 17:04.44 | paulc | praet: I <3 my SPA-3000 - it brings my building door entry system to my Polycom phone, and cell phone for when I'm not home. Ask away and I'll try and help |
| 17:05.00 | slidesinger | Been looking on google to see if there is an asterisk command that will allow to import extensions from a spreadsheet or from csv. Keeps turning up a fledgling freepbx plugin (bulk extension) that I do not wish to use outside of the lab at this time. Does such a command/tool exist or am I wasting my time? |
| 17:05.16 | praet | well I get sip peers for both 1-pstn and the extension, but on an incoming call, it picks up, i hear less than 1sec of static then it drops |
| 17:06.13 | paulc | slidesinger: you could write yourself a quicky script (Perl, PHP, whatever) to parse your CSV into the right format for sip.conf and extensions.conf |
| 17:06.13 | praet | also, in testing outgoing calls, it gets routes to the spa, but then i hear dialtone, and dtmf dialing wont work (the number called is somehow not dialed) |
| 17:06.13 | *** join/#asterisk tacubo (n=tacvbo@189.146.188.184) |
| 17:06.18 | paulc | praet: hmm.. incoming call on the FXO passing to the FXS? or SIP to FXS? |
| 17:06.19 | p3nguin | praet: Do you mean "both 1-pstn and the line 1"? |
| 17:06.20 | slidesinger | paulc: I like the sound of that. |
| 17:06.32 | praet | in the spa status page: PSTN Peer Number: the number i try to dial |
| 17:06.34 | [TK]D-Fender | slidesinger: There is no * command to import ANYTHING |
| 17:06.37 | praet | yes pstn and the line 1 |
| 17:06.42 | [TK]D-Fender | sliand FreePBX is NOT supported here |
| 17:06.44 | [TK]D-Fender | ~freepbx |
| 17:06.45 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 17:06.54 | praet | incoming call on the FXO -> asterisk |
| 17:06.57 | paulc | slidesinger: I'd build a template file then use the replace functions in PHP to "do the right thing" |
| 17:07.02 | praet | the asterisk log shows nothing |
| 17:07.09 | *** join/#asterisk Godfather_ (n=Godfathe@79.109.251.250.dyn.user.ono.com) |
| 17:07.44 | p3nguin | praet: If the incoming call from the PSTN (on the FXO) is being sent into Asterisk, then it sounds like you have the PSTN-to-VoIP Gateway setting enabled. |
| 17:08.33 | slidesinger | paulc: thanks. I don't use freepbx anywhere except the lab at this time. I don't trust GUIs until I know what they do and how they do it. |
| 17:08.36 | *** join/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
| 17:08.44 | praet | p3nguin: yes i setup PSTN-To-VoIP Gateway Setup to point to the asterisk box |
| 17:08.57 | p3nguin | praet: That's why the call goes to Asterisk, then. |
| 17:09.22 | p3nguin | praet: Increase the answer delay, enable ring-thru. |
| 17:09.38 | praet | p3nguin: oh sorry.. I _am_ trying to get the call to go to asterisk.. What happens is the call is nearly immediatly dropped |
| 17:11.18 | p3nguin | oh hmm |
| 17:11.19 | praet | the FXo port picks up, then hangs up |
| 17:11.19 | bmoraca_work | praet: pastebin a sip debug from asterisk with the failed call. that'll tell us why the call is being dropped |
| 17:11.19 | bmoraca_work | ~pastebin |
| 17:11.20 | infobot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste |
| 17:11.20 | bmoraca_work | no infobot? QQ |
| 17:11.21 | p3nguin | ~pb |
| 17:11.21 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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 apt-get install pastebinit |
| 17:11.22 | p3nguin | It was here 4 minutes ago. |
| 17:11.28 | p3nguin | lagged |
| 17:11.35 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:18e1:5759:a077:acde) |
| 17:11.41 | cusco | hi |
| 17:11.44 | bmoraca_work | way lagged, wow |
| 17:11.51 | cusco | asterisk is memory leaking |
| 17:12.06 | cusco | bmoraca_work: nickserv is |
| 17:12.08 | Chainsaw | hands cusco a beautiful blue bit bucket |
| 17:12.08 | bmoraca_work | cusco: better get some duct tape :) |
| 17:13.22 | bmoraca_work | i wonder what happens when the unique channel identifier in 1.6.2 gets up to FFFFFFFF...does it just roll back to 00000000? i'm only at 000086e2, so I've got some time, but maybe asterisk just blows up or something...curious |
| 17:13.22 | paulc | I'm really not in the mood for this training session today.. Nexidia IS interesting, but it's a bit slow going |
| 17:13.46 | cusco | looking at pmap it is growing stuff like > 00000000023b7000 853784K rw--- [ anon ] |
| 17:14.31 | cusco | all that [ anon ] |
| 17:14.47 | cusco | anyway it uses about 10% more memory everyday |
| 17:14.48 | cusco | :( |
| 17:14.50 | *** join/#asterisk albertoandrade (n=albertoa@189.114.186.118) |
| 17:14.50 | cusco | its a pain |
| 17:15.19 | praet | well it never 'gets' to asterisk on the FXO incoming, but here is the outbound log: http://asterisk.pastebin.com/d57516c00 |
| 17:15.24 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 17:16.18 | cusco | also asterisk process has LOADS of childs |
| 17:22.01 | paulc | praet: So do you want the FXO call to go to Asterisk, or the local phone, or both? |
| 17:22.16 | praet | i only need it to go from FXo into asterisk |
| 17:23.15 | praet | paulc: how do you have it configured? |
| 17:23.33 | p3nguin | No phone on line1 jack? |
| 17:23.41 | paulc | praet: That's exactly how I've got mine set up - so my door entry phone goes to Asterisk then out to my Polycom and/or cell. Let me see if I can get a screen shot. |
| 17:24.13 | paulc | Yeah, I've got a phone on there too but it doesn't ring through.. it goes through my structured cabling to a POTS phone in another room |
| 17:24.36 | bmoraca_work | praet: i don't see any problems in that debug...what is it you're trying to do and what's not working? We need a debug of a FAILED call, not a successful one |
| 17:24.39 | praet | does there need to be a phone plugged in? |
| 17:24.54 | *** join/#asterisk Tim_Toady (n=moi@188.4.5.122.dsl.dyn.forthnet.gr) |
| 17:24.59 | p3nguin | You don't have to have a phone on line1. |
| 17:25.11 | p3nguin | You can disable line1 if you don't want to use a phone on it. |
| 17:25.41 | praet | bmoraca_work: thanks for looking.. That call got connect from a softphone out the FXO port, but it didnt send dtmf codes to dial out the pstn line.. So i hear dialtone, then 'if you'd like to make a call...' |
| 17:26.08 | bmoraca_work | praet: that's a configuration issue with your ATA |
| 17:26.24 | bmoraca_work | praet: asterisk isn't going to send any DTMF in this instance. your ATA is. |
| 17:27.16 | praet | ok |
| 17:27.23 | bmoraca_work | praet: additionally, you're sending a 9 to your ATA. i suspect you want to strip that off before the ATA gets it...unless you're connecting this to another PBX that requires a 9 to dial out |
| 17:28.10 | praet | bmoraca_work: yes this is connectiong to an office pbx that requires 9 for outside line |
| 17:28.20 | bmoraca_work | ok |
| 17:28.30 | praet | I think i need to reset this spa and try again |
| 17:28.40 | praet | or get syslog debugging working |
| 17:29.01 | *** join/#asterisk RobH (n=robh@rob.tech.wikimedia.org) |
| 17:29.04 | bmoraca_work | your issue, then, is between the ATA and that PBX. asterisk is properly placing the call to the ATA. |
| 17:29.58 | praet | ok so outgoing from asterisk to the ATA is 'working'. I'm looking at settings now to see why Im not getting anything in asterisk from internal pstn calls |
| 17:30.16 | p3nguin | I thought the problem was when a call came from the PSTN into the ATA, which was supposed to go to Asterisk. |
| 17:30.36 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 17:30.38 | p3nguin | PSTN-to-VoIP gateway |
| 17:30.41 | bmoraca_work | praet: we need to see a sip debug of that failed scenario. |
| 17:30.50 | bmoraca_work | p3nguin: he's having problems both ways |
| 17:32.00 | p3nguin | praet: You mentioned that you have a phone connected to the ATA, but not to the ATA's line1 jack (which is the jack for a phone). Did I understand you correctly? |
| 17:32.36 | *** join/#asterisk Izinucs (n=Quassel@ip72-197-240-157.sd.sd.cox.net) |
| 17:33.51 | Izinucs | Is there a generic voip or IP-PBX channel? |
| 17:34.47 | [TK]D-Fender | Izinucs: Nope, this tends to field most of those people |
| 17:36.47 | Izinucs | [TK]D-Fender: thanks.. it is a generic question.. Business DSL from AT&T seems to be blocking outbound smtp traffic from my voip server. Probably see's it as spam. AT&T doesn't know anything about Smart Host in either Level 1 or 2. Are there any docs on how to get this fixed by either me or att or a combination? |
| 17:37.22 | [TK]D-Fender | Izinucs: So far that isn't even a VoIP or PBX question at all |
| 17:37.23 | p3nguin | Use AT&T's outbound relay. |
| 17:37.25 | Qwell | That's not a voip questiona t all O.o |
| 17:37.27 | [TK]D-Fender | Izinucs: qPoor lead-in |
| 17:37.54 | Izinucs | [TK]D-Fender: well it's traffic from my IP-PBX voice mail to email function.. |
| 17:37.56 | bmoraca_work | Izinucs: have you verified with your receiving SMTP server that the message isn't just getting dropped there? |
| 17:38.02 | [TK]D-Fender | Izinucs: doesn't matter |
| 17:38.11 | praet | p3nguin: there is no phone plugged in the 'phone' jack, my pots line plugged into 'line' jack, and lan power connected |
| 17:38.36 | [TK]D-Fender | Izinucs: Don't complain to Ford that you don;t like the programming a radio station director arranges just because you listen to it on the radio included in your car |
| 17:39.27 | p3nguin | Try configuring your relay host to smtp1.attglobal.net |
| 17:40.48 | bmoraca_work | p3nguin: he shouldn't have to relay on an AT&T business account. they only block outbound port 25 on residential accounts. though we have no proof that he's actually ON a business account...and they'll remove the block if you ask them and have a legitimate reason |
| 17:41.22 | [TK]D-Fender | <3 's his home ISP that give him free-reign and loves that I run *nix |
| 17:41.31 | Izinucs | bmoraca_work: from what att says the domain that I use in the pbx is being blocked because of Spamcannibal listing of the domain.. but spamcannibal lists almost all ISP's that way.. Thus allowing only verified traffic to be relayed. The chore is getting them to tell me an appropriate method of authentication. |
| 17:42.02 | Naikrovek | [TK]D-Fender: we don't get that in the US. |
| 17:42.38 | p3nguin | We don't? |
| 17:42.40 | Izinucs | [TK]D-Fender: please.. It's a client and not my account.. It must have been an issue with Asterisk's ability to do voice mail to email and people on AT&T DSL.. |
| 17:42.45 | chuckf | has that in the US |
| 17:42.48 | p3nguin | me too |
| 17:43.12 | Izinucs | p3nguin: I'll give that a shot.. I've already had them open port 25.. |
| 17:43.15 | bmoraca_work | Izinucs: irrelevant. unless you're sending to a domain that actually looks that stuff up, it doesn't matter. have you configured your receiving mailserver to do reverse DNS or SPF lookups? all of those things could cause the message to be dropped and the only way to check that is to look at the logs on your receiving system. |
| 17:43.35 | [TK]D-Fender | Izinucs: Asterisk has JACK SHIT to do with TRANSMITTING e-mail |
| 17:43.41 | [TK]D-Fender | izathats up to the MTA that you use |
| 17:44.23 | [TK]D-Fender | Izinucs: go figure out which one you're using and go read its instructions for setting the outbound proxy |
| 17:45.01 | Izinucs | bmoraca_work: apparently ATT does look that stuff up.. |
| 17:45.36 | bmoraca_work | Izinucs: no, they don't. they don't scan messages that are not directly sent to them. why would they bother? they're just the transport. |
| 17:46.01 | bmoraca_work | Izinucs: Teir 1 and Teir 2 assume you mean that you're sending to their servers, which is why they would have mentioned that. |
| 17:46.10 | Izinucs | [TK]D-Fender: all you gotta say is that my questions would be better answered in #<somewhere else> and leave it at that.. instead of responding like you have a stick up you ass. |
| 17:46.38 | bmoraca_work | Izinucs: until you check the LOGS on the RECEIVING server to VERIFY that the messages are not being dropped there, you cannot be helped further. |
| 17:46.56 | [TK]D-Fender | Izinucs: You seemed to insist that * has something to do with this. I also don't know which #whatever supports your "whatever" |
| 17:47.42 | Izinucs | bmoraca_work: the email that is outbound is coming back into the business using att email that the employees use.. so you are right.. they do look at it on the inbound side for sure. Most likely that's where it's getting dropped.. thanks.. you've helped me in my path to resolution.. |
| 17:47.45 | [TK]D-Fender | Izinucs: You also asked about a generic VoIP & PBX channel for your questions and now allude that its actually * related in some capacity |
| 17:47.52 | *** part/#asterisk Izinucs (n=Quassel@ip72-197-240-157.sd.sd.cox.net) |
| 17:48.19 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 17:48.37 | bmoraca_work | wow |
| 17:49.16 | bmoraca_work | call them and tell them to whitelist the domain...not hard. |
| 17:51.54 | praet | bmoraca_work: you were right, the 'line' to the office pbx has a crazy amount of static (its a hunted trunk so I didnt notice) Im trying to find a better suited trunk |
| 17:54.47 | *** join/#asterisk plundra (i=404@article.se) |
| 17:54.51 | praet | paulc: ok its routing into the softphone extension.. thanks for the config |
| 17:55.58 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:55.58 | bmoraca_work | blast those cursed line quality issues |
| 17:58.16 | paulc | praet: Glad to be of help :) |
| 17:58.50 | *** join/#asterisk Zaise (n=for@host191-47-static.29-79-b.business.telecomitalia.it) |
| 17:58.57 | *** join/#asterisk rgsteele (n=rgsteele@207.106.239.81) |
| 17:59.15 | rgsteele | Hey folks. Is there a way to get the dialed number via an asterisk variable, when one is using SIP? |
| 17:59.34 | Naikrovek | i wonder sometimes just how much headache i've saved myself by using a VoIP provider rather than a plain voice provider and using FXO cards |
| 17:59.38 | rgsteele | I know CALLERID(dnid) works for POTS/ISDN, but I'm looking more for something like DNIS |
| 18:00.06 | paulc | rgsteele: The number dialed by the user (or inbound destination) = ${EXTEN} ? |
| 18:00.28 | Naikrovek | that's what I was going to say: ${exten} but if i was going to say that it's probably wrong |
| 18:00.31 | *** join/#asterisk tacvbo (n=tacvbo@189.146.181.217) |
| 18:01.49 | plundra | Do any of the asterisk "distros" have an easy way of setting upp E1-interfaces? |
| 18:02.08 | Naikrovek | plundra: thsi probably isn't the place to ask, we don't know a l ot about those |
| 18:02.23 | plundra | Naikrovek: Mkay. |
| 18:02.29 | rgsteele | paulc: I mean the number the user dialed. EXTEN doesn't seem to work |
| 18:02.31 | Naikrovek | plundra: but my guess is yes |
| 18:03.15 | plundra | Naikrovek: Maybe some one could help with a regular installation instead :-) |
| 18:03.28 | plundra | It's not really Asterisk that's the problem (yet) though. |
| 18:03.43 | bmoraca_work | plundra: the only two that i know of that will aid you in setting up DAHDI or Zaptel devices are EvolutionPBX and Elastix. most others will make you set up your cards manually. |
| 18:03.43 | Naikrovek | vanilla asterisk? this is the place for that |
| 18:04.06 | paulc | rgsteele: is the user an extension off the system, or an inbound PSTN call? |
| 18:04.12 | plundra | YEah |
| 18:04.13 | plundra | bmoraca_work: Mkay. |
| 18:04.13 | cusco | hi |
| 18:04.22 | cusco | what is the most likely reaso for: |
| 18:04.23 | cusco | "[Jan 26 18:04:37] WARNING[15178]: app_queue.c:4937 queue_exec: Unable to join queue 'Covilha'" |
| 18:04.34 | plundra | Naikrovek: And it's not at asterisk yet, I think, more at the dahdi stage. But that's hear too I guess? |
| 18:04.36 | bmoraca_work | plundra: i wouldn't recommend using either if you don't know much about Asterisk. |
| 18:04.46 | cusco | unlike most queues, this one is not realtime mysql, instead its in queues.conf |
| 18:04.49 | Naikrovek | plundra: yeah |
| 18:05.00 | rgsteele | paulc: Here's the thing, using the POTS line in to the FXO card, DNIS works and such. But toll-free numbers over the SIP trunk is different - the variables seem not to be populated |
| 18:05.02 | cusco | how can I find why can't I join this queue |
| 18:05.10 | plundra | bmoraca_work: I think I've got the "regular" part of asterisk down, some what at least :-) Have a box running fine with some sip and ha ael-plan. |
| 18:05.17 | [TK]D-Fender | cusco: Where's your configs? Where is the failed call with complete CLI output? Where is the queu status dump prior to testing? |
| 18:05.20 | rgsteele | paulc: I'm talking about inbound POTS to FXO card vs. inbound SIP |
| 18:05.31 | cusco | [TK]D-Fender: ok hold |
| 18:05.47 | plundra | bmoraca_work: But this is another box at one of our offices, for which I've gotten an openvox interface (e1), which I want to use. |
| 18:06.01 | rgsteele | The inbound to POTS/FXO gives me what I'm looking for in ${DNIS}. But, when it comes in over the SIP trunk, that variable is empty |
| 18:06.06 | bmoraca_work | plundra: the dahdi installation guide is pretty repleat and easy to use |
| 18:06.08 | plundra | I might be in over my head though, isdn pri wasn't that easy :) |
| 18:06.32 | paulc | rgsteele: so which context/extension are the SIP calls landing in? usually you could get ${EXTEN} within that context |
| 18:08.25 | *** join/#asterisk jaskew (n=jdaskew@netblock-66-159-217-102.dslextreme.com) |
| 18:09.03 | rgsteele | We have some logic where if it can't figure out what the number dialed is, it sends it to a default extension. |
| 18:09.12 | rgsteele | All calls over SIP are going to that default extension. |
| 18:09.36 | cusco | [TK]D-Fender: http://paste.debian.net/57736/ |
| 18:09.38 | rgsteele | Because I can't seem to find a variable that contains the number dialed. I can see in a wireshark trace that the number dialed is actually in the SIP packets received. |
| 18:11.21 | [TK]D-Fender | gsteele: look at the rgsteele I think you should be showing us whats actually happening.. |
| 18:11.54 | paulc | rgsteele: make an extension like _x.,1,NoOp(${EXTEN} received) to see if it's being passed in correctly |
| 18:11.54 | plundra | If I have an E1 interface (opv D115P), do I still use the dahdi_dummy for timing? (It says no timing sources was found and fell back on the dummy) |
| 18:12.35 | [TK]D-Fender | cusco: You have no valid members. great reaso for it to kick you out <- |
| 18:13.01 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 18:13.02 | [TK]D-Fender | cusco: You also haven't set normal parameters like you should. Go read the sample configs |
| 18:13.13 | [TK]D-Fender | plundra: No. |
| 18:13.21 | cusco | [TK]D-Fender: dooh, right it says invalid, sorry |
| 18:13.42 | cusco | normal parameters? in queues.conf? |
| 18:13.43 | plundra | [TK]D-Fender: Ok. That's what I thought, thanks. |
| 18:13.55 | bmoraca_work | plundra: no, you don't...provided your E1 is properly configured. |
| 18:14.01 | cusco | [TK]D-Fender: I didn't paste the whole queues.conf |
| 18:14.05 | rgsteele | I have been NoOp'ing and printing out the variables - ${EXTEN} is the string "s". The first line of my inbound context, in which the SIP calls arrive, is "exten => s,1,Macro(accept-inbound-call|aweber)" - I just cant' figure out why ${EXTEN} is showing up as "s" |
| 18:14.12 | plundra | bmoraca_work: YOu mean my end or the actual connection? |
| 18:14.27 | bmoraca_work | plundra: your card, mostly |
| 18:14.33 | paulc | rgsteele: do you register to your SIP provider per DID? |
| 18:14.49 | rgsteele | paulc: We register the trunk. |
| 18:14.59 | plundra | bmoraca_work: Ok, well I wouldn't bet on anything in my end :-) |
| 18:15.18 | rgsteele | One username/password - all the toll-free's belong to the trunk. |
| 18:15.22 | [TK]D-Fender | rgsteele: It goes to "s" because you didn't tell your REGISTER to tell them otherwise |
| 18:15.51 | [TK]D-Fender | rgsteele: they should target a specific EXTEN with their calls if multiple are associated with a given registration. |
| 18:15.57 | bmoraca_work | plundra: just follow the dahdi installation instructions...it's not terribly difficult |
| 18:16.06 | [TK]D-Fender | rgsteele: Or in some freak cases you may have to manually parse out the To: header |
| 18:16.13 | plundra | bmoraca_work: I'll check them out again, yes. |
| 18:16.37 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:16.55 | rgsteele | [TK]D-Fender: That's what the registrar suggested - parsing the To: header, but I would rather not do that. I'm not sure what you mean by "tell[ing] the register to tell them otherwise"? |
| 18:17.05 | paulc | rgsteele: I register to my provider and I get the DNIS fine in my context (ie without using /xxxx on the register line) |
| 18:17.28 | paulc | it seems weird |
| 18:17.36 | bmoraca_work | paulc: it varies provider to provider. i don't require my customers to specify that, but some do |
| 18:17.38 | [TK]D-Fender | rgsteele: In your case "To:" is the way to go |
| 18:18.19 | rgsteele | [TK]D-Fender: Know of any documentation describing the process? |
| 18:18.32 | [TK]D-Fender | rgsteele: core show function SIP_HEADER |
| 18:18.36 | *** join/#asterisk Iamnach0 (n=Iamnacho@ip174-70-139-208.ks.ks.cox.net) |
| 18:18.50 | [TK]D-Fender | rgsteele: basic variable stuff. |
| 18:19.11 | rgsteele | [TK]D-Fender: Thanks! |
| 18:20.32 | paulc | bmoraca_work: does it vary on if they have one DID or multiple? if they specify /xxxx you can't tell which DID was dialed, surely? |
| 18:21.58 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:24.59 | *** join/#asterisk benngard (n=benngard@90-230-92-67-no148.tbcn.telia.com) |
| 18:25.12 | plundra | bmoraca_work: But ok, as far as the installation guide (If I'm not looking at something else http://docs.tzafrir.org.il/dahdi-linux/) goes, I think I have that down. |
| 18:25.58 | *** join/#asterisk psilikon (n=Joel@cerberus.vicimarketing.com) |
| 18:26.08 | plundra | So maybe it's the asterisk-part I am having trouble with. (I am not sure what goes where, I mean what to expect from each "step") |
| 18:26.23 | benngard | had an lucky day |
| 18:26.27 | benngard | a* |
| 18:27.06 | benngard | both getting rfp 32 to work with asterisk and manage to load stamping machines |
| 18:27.38 | *** join/#asterisk tacubo (n=tacvbo@189.146.176.217) |
| 18:28.19 | benngard | e1 - avaya - the new ooh323 driver - asterisk - spa2102 - stamping machine |
| 18:28.24 | paulc | benngard: success with your franking machines tele-loading then? via ATAs? do they use DTMF or modems? |
| 18:28.34 | benngard | se above |
| 18:28.46 | p3nguin | ${SIP_HEADER(To)0:5:10} will give me the phone number that is called, and I could use that for which exten to use... but what about when the call comes over IAX2 instead of SIP? |
| 18:29.30 | file | IAX2 doesn't have that problem. |
| 18:29.39 | benngard | paulc: if i find the address to may213 i gonna send him a bottle of "absolut vodka" ;) |
| 18:29.44 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:30.01 | p3nguin | file: Oh, okay. Issue solved, then. |
| 18:30.03 | Qwell | benngard: That could probably be arranged. |
| 18:30.18 | benngard | i will fix that in some way for him! |
| 18:31.30 | benngard | i havent checked the svn today, but i guess i have the the very latest ooh323 driver on my box |
| 18:31.49 | benngard | we were tired, me and may yesterday |
| 18:32.28 | benngard | i dont dare to do a svn up until he allows me to do so |
| 18:32.53 | Qwell | benngard: You can do `svn st -u` to see what would be updated |
| 18:33.04 | paulc | I'm all for sharing the vodka.. or beer.. Mmm beer - I could do with one to get through this training ;-) |
| 18:33.22 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 18:33.25 | benngard | Qwell: thx, didnt know that |
| 18:34.18 | cusco | [TK]D-Fender: sorry but.. I can't figure why the member is invalid. I can call him directly |
| 18:34.43 | benngard | he solved fix this: i was very impressed, a phone registered to avaya calls a phone registered to *, when sip phone answers name on avaya phone display changed from number to name! ;) |
| 18:35.13 | [TK]D-Fender | cusco: Module load issue. " |
| 18:35.34 | [TK]D-Fender | cusco: Common... do "module reload app_queue.so" and check the status again |
| 18:36.17 | cusco | [TK]D-Fender: keeps showing as invalid |
| 18:36.31 | [TK]D-Fender | cusco: complete unload/relaod |
| 18:36.40 | [TK]D-Fender | cusco: there was a small trick to this... |
| 18:37.09 | plundra | Who keeps loading chan_dahdi.so even though I have noload in modules.conf? (I want to start asterisk with all debuging on etc. and THEN load the dahdi-stuff, so the output isn't mixed up with everything else) |
| 18:37.43 | cusco | [TK]D-Fender: can't unload, there are 15 calls right now in queue bridged to operators... |
| 18:38.33 | Katty | well i'm pretty disappointed in the new webcam |
| 18:38.33 | bmoraca_work | plundra: your system.conf in /etc/dahdi and your chan_dahdi.conf in /etc/asterisk are the two files you need to look at...there are many examples on the net and the sample files are very well commented and documented...unless you have a specific error, i can't really help you. additionally, you can always reload the module. |
| 18:38.35 | benngard | Qwell: when i do execute "your" command: i got for example: M * 242809 addons/ooh323c/src/ooh323.c guess that means that "svn" wants to update that file, correct? |
| 18:38.40 | plundra | Ah, hmm, disabling autoload will do for now. |
| 18:39.15 | Qwell | M means it's modified, * means it's out of date |
| 18:39.22 | Qwell | (modified locally, that is) |
| 18:39.22 | *** join/#asterisk neurosys (n=neurosys@32.161.32.161) |
| 18:39.23 | plundra | bmoraca_work: I do have a specific error; "chan_dahdi.c:2121 dahdi_open: Unable to specify channel 1: No such device or address" |
| 18:39.28 | Katty | http://www.ustream.tv/channel/the-nut-house-bird-bath <- i guess it's okay, but it's not really doing what i want it to. i need it to zoom in so i can see MORE of what i'm wanting to see and less of what i don't care about. |
| 18:39.55 | plundra | bmoraca_work: Using the dahdi-channels.conf generated by dahdi_genconf. |
| 18:39.59 | Katty | time to buy an hd camcorder off newegg i guess |
| 18:40.26 | benngard | Qwell: thx, i guseed that, means that i will not do anything until may commited the changes |
| 18:40.43 | Qwell | good idea |
| 18:40.52 | benngard | i know that he found some issues that we are gonna take care about tonight |
| 18:40.54 | bmoraca_work | plundra: why don't you pastebin your system.conf, your dahdi-channels.conf, and the output of dahdi_config -vvvvv |
| 18:41.30 | benngard | but as i said, we where damn tired yesterday when we stopped (he programming and me testing) |
| 18:43.56 | plundra | bmoraca_work: http://pastie.org/private/iwtah7ri4cgcto0kteb1q |
| 18:44.29 | plundra | Included dahdi_scan for good measure. |
| 18:44.54 | bmoraca_work | plundra: alarms=RED/REC |
| 18:44.55 | bmoraca_work | <PROTECTED> |
| 18:45.11 | Katty | anyone wanna sell me an hd camcorder? (= |
| 18:45.26 | *** join/#asterisk Yedidya_ (i=56b4491f@gateway/web/freenode/x-rgxyuecnjknpsqsx) |
| 18:46.14 | [TK]D-Fender | Katty: that one you showd me last year was SD? |
| 18:46.16 | plundra | bmoraca_work: I guess so yes. (It alternates between red/rec and just red after failing the recovery, I assume) |
| 18:46.26 | cusco | [TK]D-Fender: I did a small change on queues.conf and reloaded, it fixed it |
| 18:46.32 | cusco | how weird |
| 18:46.46 | bmoraca_work | plundra: have you verified with a loopback that it works? also, what are you plugging in to? you may need to use a T1 crossover or roll your T1 jack |
| 18:46.50 | [TK]D-Fender | cusco: Yeah, its all about load order... autoload seems to screw up on this |
| 18:46.57 | plundra | bmoraca_work: So it ought to be a physical issue? Right. |
| 18:46.57 | bmoraca_work | by "works" i mean turns green |
| 18:47.08 | Katty | [TK]D-Fender: hmm? |
| 18:47.09 | plundra | bmoraca_work: I haven't got a loopback or any other cable. |
| 18:47.15 | bmoraca_work | plundra: potentially, yes. i don't see anything wrong with your configs |
| 18:47.29 | plundra | bmoraca_work: I'm pluggin in to the outlet provided by our provider. |
| 18:47.33 | [TK]D-Fender | Katty: I recall your shopping for a camera, one was a metallic red panasonic one |
| 18:47.33 | cusco | [TK]D-Fender: where can I find configs for autoload ? |
| 18:47.36 | bmoraca_work | plundra: do you have any spare RJ-45 connectors? |
| 18:47.39 | [TK]D-Fender | cusco: modules.cong |
| 18:47.40 | [TK]D-Fender | f |
| 18:47.47 | cusco | thanks |
| 18:47.49 | *** join/#asterisk korihor (n=korihor@201.210.226.98) |
| 18:48.02 | plundra | bmoraca_work: I'll just slaughter a cable for it, so yes. |
| 18:48.28 | rgsteele | [TK]D-Fender: One more quick question - is "s" the default when no inbound destination number is known? |
| 18:48.32 | plundra | bmoraca_work: But I'll start with se if I can some how lirk the cable going to the current PBX out of it's nest... |
| 18:48.42 | [TK]D-Fender | rgsteele: No. |
| 18:48.42 | rgsteele | In all cases? |
| 18:48.43 | cusco | [TK]D-Fender: can I tell a certeain module to be loaded/reloaded after autoload in that conf |
| 18:48.46 | [TK]D-Fender | ~stdextens |
| 18:48.46 | infobot | [~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
| 18:48.48 | cusco | ? |
| 18:48.58 | [TK]D-Fender | rgsteele: Actually... YES, sorry |
| 18:49.06 | [TK]D-Fender | rgsteele: In your case it IS known.. and it is "s" |
| 18:49.16 | rgsteele | [TK]D-Fender: Heh, okay, cool. |
| 18:49.26 | rgsteele | When you said no, I was a bit confuzzled for a sec :) |
| 18:49.39 | Katty | [TK]D-Fender: oh. |
| 18:49.39 | bmoraca_work | plundra: have you also confirmed that your framing and line coding are correct for the circuit? |
| 18:49.39 | [TK]D-Fender | cusco: there is a series of preload directives or something of the like. I never fixed this myself before but helped someone who did |
| 18:50.25 | cusco | [TK]D-Fender: I guess the preload will load that before the autoload... so I am guessing I should load it after...? |
| 18:50.28 | Katty | [TK]D-Fender: i want something better than that. |
| 18:50.38 | Katty | [TK]D-Fender: if i'm going to get a new camcorder, i'm going to get an HD one |
| 18:51.05 | [TK]D-Fender | Katty: You;re not about to stream HD off this I'm sure... You need a better lens, etc.. you COULD probably use your camcorder as a direct cam sourse |
| 18:51.21 | p3nguin | looks at his old handycam and sighs |
| 18:51.25 | Katty | my camcorder sucks. |
| 18:51.29 | Katty | it's from last decade. |
| 18:51.40 | Qwell | the 90s? |
| 18:51.45 | Katty | late 90s, yes. |
| 18:51.46 | cusco | december 2009 |
| 18:51.47 | cusco | :p |
| 18:51.53 | cusco | (since we are in 2010) |
| 18:51.54 | Qwell | cusco: That would be current decade |
| 18:52.13 | Katty | it records to TAPE |
| 18:52.16 | p3nguin | 2000 would be the end of hte last decade. |
| 18:52.17 | cusco | hmm 2010 - 10 = 2000 |
| 18:52.17 | Katty | DUNDUNDUN |
| 18:52.23 | cusco | woa |
| 18:52.37 | cusco | does you asterisk box also records to tape? :P |
| 18:52.46 | Katty | no, it records to stone. |
| 18:52.49 | p3nguin | katty: To VHS? |
| 18:52.51 | bmoraca_work | that'd be a neat trick |
| 18:53.04 | cusco | bmoraca_work: hehe, next it will show up on flinstones |
| 18:53.06 | Kobaz | Katty: http://www.thinkgeek.com/electronics/portable-audio-video/c773/ |
| 18:53.07 | benngard | flintstone ;) |
| 18:53.12 | Qwell | cusco: 2000 was also not a new decade |
| 18:53.24 | cusco | a decade is 10 years |
| 18:53.31 | Kobaz | Qwell: depends when you start counting |
| 18:53.31 | cusco | century 100 |
| 18:53.33 | Katty | 8 TRACK |
| 18:53.37 | Qwell | And on what year do decades start? |
| 18:53.39 | p3nguin | Mine records to tape, but it's a Digital8 tape. |
| 18:53.42 | plundra | bmoraca_work: I tried asking our provider for some good specs, but all he told me was the bchan 1-15 + 17-31 and dchan 16. |
| 18:53.46 | Qwell | What year did the first decade start? |
| 18:53.52 | cusco | ok plus 1 |
| 18:53.54 | cusco | :/ |
| 18:53.57 | Qwell | quite |
| 18:54.02 | cusco | so next year will be a new decade |
| 18:54.05 | Katty | are we really having a conversation about when a decade starts? |
| 18:54.05 | cusco | ok.. |
| 18:54.05 | Qwell | correct |
| 18:54.07 | Katty | srsly. |
| 18:54.09 | Qwell | Katty: yes |
| 18:54.17 | plundra | Not sure if the cable did anything, it's just REC now. But I believe I've had that before. |
| 18:54.20 | cusco | I was told that before, I just wanted to forget it for argument's sake |
| 18:54.22 | benngard | can someone explain for me why asterisk send connect info when a call is answered and why not on ringing? |
| 18:54.32 | Katty | Kobaz: the point isn't HD...specifically |
| 18:54.37 | Katty | Kobaz: the big thing is ZOOM QUALITY |
| 18:54.50 | cusco | benngard: that is self explanatory. It might ring but not be answered, thus not connected |
| 18:54.51 | Katty | Kobaz: i have eleventy billion feeders i need to zoom to, and broadcast, retaining high quality. |
| 18:54.58 | bmoraca_work | plundra: i'm not that familiar with E1, so I don't know what the conventional framing/coding types are...here it's esf/b8zs...but you euros are weird |
| 18:54.59 | cusco | it might for instance, try and ring somebody else |
| 18:55.08 | cusco | only connected when it won't ring again |
| 18:55.12 | voipmonk | hdb3 ccs |
| 18:55.19 | Katty | Kobaz: something tells me that pocket device isn't really awesome in the zoom department |
| 18:55.27 | voipmonk | check the sample file |
| 18:55.39 | plundra | bmoraca_work: I bet we are. T1 have some combined signaling and payload aswell, right? |
| 18:56.15 | benngard | cusco: u could send to cennect info, 1:st connect info for the phone u are trying to reach and another connect info i for example a divert happens |
| 18:56.21 | p3nguin | plundra: CAS T1 integrates them. |
| 18:56.27 | p3nguin | plundra: PRI does not. |
| 18:56.29 | bmoraca_work | plundra: we have both CCS styles and CAS styles, yes...and our CAS robs bits from the payload to do signalling whereas your CAS has a dedicated channel for signalling (so it's not really CAS)...but whatever :) |
| 18:56.35 | plundra | Another thing, the kernel is spewing out "timing source auto card 0!" |
| 18:56.51 | cusco | benngard: I guess that only changes the devicestate to busy or ringing or whatever... |
| 18:56.57 | Katty | http://www.newegg.com/Product/Product.aspx?Item=N82E16830120313 <- i'm thinking about this one |
| 18:56.59 | bmoraca_work | plundra: any green or yellow yet or back to red? |
| 18:57.15 | bmoraca_work | plundra: you should know that openvox cards aren't well loved by many |
| 18:57.24 | benngard | cusco: nop it changes the display of the caller, we saw it last night |
| 18:57.27 | plundra | bmoraca_work: No colors, just REC actually. |
| 18:58.25 | bmoraca_work | plundra: has your provider confirmed that it's live? |
| 18:58.33 | plundra | I did find some lowlevel documents from my provider, but it's 10 separate documents; so, what should I be looking for? :) Framing and coding? |
| 18:58.57 | plundra | bmoraca_work: Right now with my E1 card plugged in? |
| 18:59.02 | bmoraca_work | your framing and coding are probably right. red alarm is on your side. |
| 18:59.21 | bmoraca_work | plundra: no..have you confirmed that the provider has turned the port up? they may not have yet. |
| 18:59.22 | rgsteele | [TK]D-Fender: How do I tell * "I know the inbound number now.", so that it can match against it with exten directives? |
| 18:59.26 | benngard | pretty neat feature if i may say, for example, i call 976 (Jimmy) my display says Jimmy as long as it is ringing, he dont answer so the calls divert to 972 (Kerem) my display updates to Kerem and so on |
| 18:59.45 | plundra | bmoraca_work: Oh, yeah they have, it's working fine with our old Samsung pbx. |
| 18:59.57 | plundra | bmoraca_work: I'm here off-hours now, trying to get this thing going :) |
| 19:00.12 | *** join/#asterisk helloritesh_ (n=hellorit@121.243.172.188) |
| 19:00.19 | plundra | Have been working fine for multiple years too, so it's very much alive. |
| 19:00.23 | paulc | benngard: that's nice.. COLP - not seen a whole bunch |
| 19:00.29 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.172.179.dsl.dyn.forthnet.gr) |
| 19:00.51 | rgsteele | [TK]D-Fender: Nevermind, sorry, stupid question. |
| 19:00.52 | plundra | Oh, I'm seeing some RED/REC again, not just REC. (Running dahdi_scan every now and then) |
| 19:00.58 | rgsteele | I figured it out. |
| 19:01.28 | *** join/#asterisk flohack (n=fhackenb@chello084115131198.3.graz.surfer.at) |
| 19:01.46 | bmoraca_work | plundra: can you make a loopback plug and plug it in to see if the card even works properly? |
| 19:02.03 | plundra | I'll try that. |
| 19:02.57 | bmoraca_work | loop pin 1 to 5 and 2 to 4 |
| 19:03.49 | bmoraca_work | that's not right |
| 19:03.58 | bmoraca_work | 2 to 5, 1 to 4 |
| 19:06.29 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:08.07 | [TK]D-Fender | rgsteele: ? |
| 19:12.37 | *** join/#asterisk JraNil (n=JraNil@85.133.218.151) |
| 19:13.30 | *** join/#asterisk vader-- (n=me@c-68-36-9-8.hsd1.nj.comcast.net) |
| 19:13.32 | vader-- | hello |
| 19:13.51 | vader-- | do you guys know if a Dell Poweredge 2850 2U will handle a digium TDM2400P card? |
| 19:14.27 | *** join/#asterisk shin-shoryuken_ (n=shin-sho@hephaestus.restechservices.net) |
| 19:14.48 | [TK]D-Fender | vader--: Measure it.. but I doubt it. |
| 19:14.58 | [TK]D-Fender | vader--: Why are you looking at that card exactly? |
| 19:15.04 | vader-- | i already have it |
| 19:15.12 | vader-- | it's in a 2800 which is a larger server |
| 19:15.13 | *** join/#asterisk rare1980_ (n=rare@203.175.76.221) |
| 19:15.16 | *** join/#asterisk lanning (n=lanning@208.87.235.224) |
| 19:15.19 | [TK]D-Fender | vader--: HUGE. |
| 19:15.24 | vader-- | ya hehe |
| 19:15.45 | [TK]D-Fender | vader--: Where would you be getting the 2850 from? |
| 19:15.52 | vader-- | i already have it |
| 19:16.01 | plundra | bmoraca_work: Yay, using a crossover: alarms=OK |
| 19:16.06 | rare1980_ | hi all.. please any one who can guide me that how can i use SIP trunks use IP authentication settings |
| 19:16.21 | [TK]D-Fender | vader--: If you have the card...and you have the server.... F-ING PLUG IT IN! :P |
| 19:16.21 | plundra | bmoraca_work: And I see the kernel spit out: Opvxd115: Clearing yellow alarm on span 1 |
| 19:16.33 | bmoraca_work | plundra: good. restart asterisk and you should be able to place calls now. |
| 19:17.23 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:17.25 | plundra | bmoraca_work: Hmm, do what? :) With the loopback? |
| 19:17.35 | bmoraca_work | oh |
| 19:17.37 | rare1980_ | hello |
| 19:17.57 | voipmonk | hello |
| 19:18.01 | fenrus | hello |
| 19:18.09 | vader-- | tkd the card is in a production system |
| 19:18.19 | [TK]D-Fender | waits for the other couple hundred people to all respond "hello" |
| 19:18.21 | bmoraca_work | plundra: you said with crossover, not loopback. that means that your cable is likely not correct |
| 19:18.31 | rare1980_ | please any one who can guide me that how can i use SIP trunks use IP authentication settings |
| 19:18.32 | [TK]D-Fender | vader--: Open it up and get a ruler |
| 19:18.36 | benngard | hello :) |
| 19:18.43 | [TK]D-Fender | vader--: the odds of that combo in use by someone here = slim |
| 19:18.44 | rare1980_ | greg |
| 19:18.45 | plundra | bmoraca_work: Oh, hmm, I'm tired... Sorry. |
| 19:18.51 | voipmonk | waves |
| 19:18.52 | *** join/#asterisk eharris (i=eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 19:18.58 | bmoraca_work | plundra: or your framing/coding are not correct |
| 19:19.23 | plundra | bmoraca_work: Yeah, and the best thing is, the old PBX doesn't come up as it is supposed to :-) |
| 19:19.38 | plundra | But I'll try looking for anything mentioning coding or framing in the docs I found. |
| 19:19.53 | bmoraca_work | plundra: at that point, i'd really start to suspect the cable. |
| 19:21.58 | plundra | Maybe I'll give that a go, now when I have a cut up cable I'll just make it a crossover one. Can't be that picky with untwisted pairs etc? :) |
| 19:22.10 | *** join/#asterisk e-jones (n=jkastner@84.242.102.36) |
| 19:22.41 | bmoraca_work | plundra: well...crossover won't work when a straight-through is expected... |
| 19:23.44 | plundra | I know, I know. But as you said, maybe it's the cable? And if my straight one isn't working, making a crossover one might. |
| 19:24.01 | Katty | pouts |
| 19:24.02 | hardwire | meh.. can the 7960/40s load their SIPxxx.cnf over HTTP? |
| 19:24.05 | Katty | still no visitors to the birdy bath :< |
| 19:24.12 | Chainsaw | hardwire: Just TFTP. |
| 19:24.13 | hardwire | I'm guessing no.. can't find a single reference |
| 19:24.21 | hardwire | Chainsaw: that's great for security.. :) |
| 19:24.29 | hardwire | considering I'd love to use HTTPS |
| 19:24.33 | Chainsaw | hardwire: Yes. Just like they won't do HTTPS either :) |
| 19:24.37 | Naikrovek | Katty: birds gotta case the joint first. make sure there are no cats hanging around |
| 19:24.40 | hardwire | Chainsaw: annoying |
| 19:24.40 | Chainsaw | hardwire: Or SIP over TCP, SRTP, etc. |
| 19:24.52 | Katty | Naikrovek: it's in the same tree they hang out in every day |
| 19:24.54 | hardwire | Chainsaw: oh well. at least the linksys SPA can use HTTPS |
| 19:25.03 | Naikrovek | Katty: ah - well, patience then |
| 19:25.06 | Katty | Naikrovek: maybe it's too cold out for a drink |
| 19:25.12 | Naikrovek | maby |
| 19:25.14 | Naikrovek | maybe |
| 19:25.15 | bmoraca_work | plundra: if your straight one doesn't work in the original PBX either, there might be some breakage. you'll need to try them both with new cables. |
| 19:25.25 | Katty | Naikrovek: i decided on a camcorder (= |
| 19:25.30 | Naikrovek | Katty: they have routines i think. |
| 19:25.33 | Katty | Naikrovek: none of the stores around here carry it tho :/ |
| 19:25.34 | Naikrovek | camcorder |
| 19:25.37 | Naikrovek | oh |
| 19:25.54 | Katty | i know. having to wait a week... DOOM |
| 19:25.56 | plundra | bmoraca_work: Oh, no, I am using the original cable for the PBX again I'm suspecting it lost the config or whatever. That was an unrelated side note. |
| 19:26.15 | vader-- | tdk the 2850 was our exchange server and we just P2Ved it, which left the physical hardware open. The server has a support contract on it still. The server that asterisk is currently running (Dell Poweredge 2800) on doesn't have a support contract, we were looking at renewing the service contract but it's going to be a few grand. |
| 19:26.25 | Naikrovek | can't make his playstation bluetooth headset sit comfortably on his ear |
| 19:26.34 | bmoraca_work | plundra: so the original PBX is up and green again? if not, I'd STRONGLY suspect the cable is bad |
| 19:26.43 | vader-- | so i figured if we can move over to the 2850 and not have to pay the support contract that would work in our favor incase any issues arose |
| 19:26.49 | Katty | Naikrovek: duct tape. |
| 19:26.58 | Naikrovek | Katty: not comfortable though |
| 19:27.05 | Naikrovek | it stays on there well enough, just hurts |
| 19:27.11 | Katty | does it pinch |
| 19:27.34 | Naikrovek | wouldn't call it a pinching, it just manhandles my hear until my ear does what it wants |
| 19:27.52 | Katty | ohisee |
| 19:28.05 | Naikrovek | need to bend this thing until it fits my ear or something |
| 19:28.12 | plundra | bmoraca_work: Not sure if we're misunderstanding each other here now; 1) The cable used by the old PBX didn't do any difference in my setup. 2) The old PBX not coming up again or whatever is another problem. |
| 19:28.13 | bmoraca_work | vader--: meh...i just use used ProLiant DL360 G4s or DL380 G3s...haven't met one yet not powerful enough to handle our load...also, the vendor we get them from gives us a 3 year warranty on parts...fuckers are built like tanks |
| 19:28.53 | bmoraca_work | plundra: i know you say it's another problem, but it may be the same problem....that's my point. it's easy to check a cable. |
| 19:29.08 | Katty | Naikrovek: Canon Vixia HF200 |
| 19:29.23 | Katty | Naikrovek: it's on sale at newegg for 549ish |
| 19:29.26 | plundra | bmoraca_work: What cable exactly? :) The one from the walljack to the pbx/computer? |
| 19:29.34 | bmoraca_work | plundra: yes |
| 19:29.43 | Katty | Naikrovek: won't break the bank just for a hobby |
| 19:30.05 | Naikrovek | Katty: nice |
| 19:30.06 | *** join/#asterisk mateu (n=hunter@c-98-220-12-97.hsd1.in.comcast.net) |
| 19:30.22 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-40-42.acanac.net) |
| 19:30.24 | plundra | bmoraca_work: I really doubt I broke THAT cable when unplugging it from the PBX :-) But sure, I could try another one there. |
| 19:30.31 | Katty | i've already dumped several hundred into the front yard... |
| 19:30.46 | Naikrovek | all for the animals |
| 19:30.49 | bmoraca_work | plundra: if it's old and made out of solid core, it very well could have been damaged very easily |
| 19:30.54 | Katty | all for me |
| 19:31.01 | Katty | it's my way of dealing with the winter |
| 19:31.20 | plundra | bmoraca_work: No it's a flexible patchcable, cat5e or whatever. |
| 19:31.32 | bmoraca_work | Katty: there are other things to do during the winter to keep warm... |
| 19:31.35 | plundra | I just tried dialing out through it and that's fine, so can't be that. |
| 19:31.41 | Katty | bmoraca_work: it's not about keeping warm. |
| 19:31.51 | Katty | bmoraca_work: the house is plenty warm (= |
| 19:31.56 | bmoraca_work | ok, then, to keep occupied :) |
| 19:32.02 | plundra | bmoraca_work: So I'll continue making a crossover. |
| 19:32.04 | Katty | it's not about that either. |
| 19:32.05 | bmoraca_work | both literally and figuratively |
| 19:32.47 | bmoraca_work | plundra: that'd be another try. short of that, my only guess would be framing and coding...red alarm means it's on your side generally |
| 19:33.03 | *** join/#asterisk adam05634 (n=adam0563@173-14-238-70-Utah.hfc.comcastbusiness.net) |
| 19:33.39 | plundra | bmoraca_work: Yeah. But I'll try the cable, then go with the coding/framing-angle. Making a cable is much more fun :-P |
| 19:34.32 | Katty | i guess keeping occupied has something to do with it tho |
| 19:34.37 | Katty | and there's a cardinal eying it |
| 19:34.45 | Katty | ohoh, a branch closer :> |
| 19:35.05 | Katty | :< |
| 19:36.17 | *** join/#asterisk albertoandrade (n=albertoa@187.59.20.61) |
| 19:37.16 | adam05634 | I need some help configuring a TDM808E card with Asterisk. I have installed dahdi and asterisk and did the samples just so I could see if it worked, and the system can see my card and everything, the phonel line is plugged in and I start up asterisk but even when I set the verbosity high I cannot see the call come in when I call from the outside. Is there something that I'm missing? |
| 19:37.54 | Katty | well if it's not seeing incoming calls, i would definately say Yes, something is missing. |
| 19:38.24 | [TK]D-Fender | adam05634: "did the samples"? |
| 19:38.44 | adam05634 | I installed the sample config files |
| 19:38.54 | [TK]D-Fender | adam05634: Samples are not usable for anything |
| 19:39.10 | rgsteele | [TK]D-Fender: Got it all working - thanks again for the SIP_HEADER tip, much appreciated. |
| 19:39.15 | adam05634 | even without the samples, shouldn't I see the call come in the CLI in asterisk? |
| 19:39.44 | p3nguin | Without... WHAT? Without config files, nothing can work. |
| 19:39.56 | p3nguin | And using the samples, nothing WILL work. |
| 19:40.03 | adam05634 | okay I mean without default dial plan |
| 19:40.09 | [TK]D-Fender | adam05634: No, you would not |
| 19:40.16 | p3nguin | THere is no default dial plan. |
| 19:40.25 | [TK]D-Fender | adam05634: If * is not CONFIGured to loko at those channels than you see nothing about them. |
| 19:40.34 | adam05634 | there is a dialplan that says congratulations |
| 19:40.39 | adam05634 | at least it did last time I set asterisk up |
| 19:40.41 | [TK]D-Fender | adam05634: And there is no "default dialplan either". Samples = junk. |
| 19:40.54 | [TK]D-Fender | adam05634: For conceptual reference only |
| 19:40.56 | dlynes_laptop | ~crittercam |
| 19:40.57 | infobot | well, crittercam is Katty's broadcast of The Nut House @ http://ustre.am/8H5d |
| 19:41.53 | dlynes_laptop | Katty, those two things dangling that looks like snowflake paper sculptures or something are your new fruit thingies for the birdies? |
| 19:41.56 | plundra | bmoraca_work: Ok, so no-go on the crossover. (And I realized something I probably should have mentioned; when nothing is plugged in to the card it flashes red. But then plugging it on to the wall, it stop. It did NOT with the crossover) |
| 19:42.27 | bmoraca_work | plundra: that means you don't need a crossover |
| 19:42.41 | *** join/#asterisk simplydrew (n=simplydr@pool-74-97-177-245.prvdri.fios.verizon.net) |
| 19:43.25 | *** join/#asterisk roni (n=roni@190.196.71.206) |
| 19:43.33 | Katty | dlynes_laptop: that is plasticy, sparkly, snowflake garland |
| 19:43.49 | Katty | infobot: critter |
| 19:43.50 | infobot | Perl/Tk flat text search script. URL: http://www.mainmatter.com/critter/ |
| 19:43.50 | adam05634 | do I need a dialplan for incoming to see the call coming in on the Asterisk CLI? |
| 19:43.56 | Katty | infobot: critter cam |
| 19:43.57 | infobot | it has been said that critter cam is Katty's broadcast of The Nut House @ http://ustre.am/8H5d |
| 19:44.02 | roni | hi everybody |
| 19:44.15 | Katty | infobot: forget critter cam |
| 19:44.16 | infobot | i forgot critter cam, Katty |
| 19:44.21 | plundra | bmoraca_work: And about framing/codec, is something more important? Or do I really need the correct combo for the alarm to go away? |
| 19:44.24 | Katty | infobot: forget crittercam |
| 19:44.24 | infobot | i forgot crittercam, Katty |
| 19:44.42 | bmoraca_work | plundra: if you do not have the correct combo, your T1 will not work. |
| 19:45.17 | plundra | bmoraca_work: Yeah, I get that, bt I mean, say I go with a brute-force approach (I won't for now), can you first figure out the framing and THEN the codec? |
| 19:45.52 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 19:45.53 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 19:45.57 | bmoraca_work | plundra: i suspect not, though you might get yellow alarm with one right and the other wrong |
| 19:46.20 | [TK]D-Fender | adam05634: If the calls don't ahve a proper place to land, then they won't land at all and your call won't get processed |
| 19:47.13 | vader-- | so i figured if we can move over to the 2850 and not have to pay the support contract that would work in our favor incase any issues arose |
| 19:47.13 | *** join/#asterisk luke-jr (n=luke-jr@2002:62b3:1d4c:0:20e:a6ff:fec4:4e5d) |
| 19:47.18 | vader-- | tdk the 2850 was our exchange server and we just P2Ved it, which left the physical hardware open. The server has a support contract on it still. The server that asterisk is currently running (Dell Poweredge 2800) on doesn't have a support contract, we were looking at renewing the service contract but it's going to be a few grand. |
| 19:47.42 | luke-jr | http://pastebin.ca/1766959 <-- any ideas on this segfault? :| |
| 19:49.42 | vader-- | tkd can you recommend a 24 port FXS device that hooks up over ethernet or usb? |
| 19:49.54 | plundra | Oh OH OH :-D |
| 19:50.12 | Katty | infobot: crittercam is Katty's broadcast of The Nut House Critter Cam @ http://ustre.am/8H5d and The Nut House Bird Bath @ http://ustre.am/bEBU |
| 19:50.13 | infobot | Katty: okay |
| 19:50.13 | bmoraca_work | vader--: mediatrix has one...Adtran has one (Atlas or Total Access) |
| 19:50.47 | plundra | bmoraca_work: I might be happy. We'll see :-) Probably fucked up something else to skew the "test", but connected to my provider alarms is OK now :-) |
| 19:51.03 | bmoraca_work | plundra: what'd you change? |
| 19:51.28 | [TK]D-Fender | vader--: Mediatris 4124/1124, AudioCodes MP-124 |
| 19:52.15 | plundra | Remvoed crc4 from the span=... in system.conf. BUT, I also might have loaded modules in a different order. But seeing as the loopback worked, I doubt that could have changed something. |
| 19:52.27 | plundra | I'll fire up Asterisk now and see what chan_dahdi says. |
| 19:53.08 | bmoraca_work | plundra: "pri show" commands are useful, too..."Provisioned, Up, Active" is what you want to see |
| 19:53.30 | plundra | Oh my, OH MY :-) -- Extension '688' in context 'from-pstn' from '73MYFREAKINGMOBILE' does not exist. Rejecting call on channel 0/14, span 1 |
| 19:53.56 | vader-- | are you guys using any of those devices? |
| 19:54.00 | bmoraca_work | plundra: there you go |
| 19:54.26 | plundra | bmoraca_work: Thank you so much for you patiance and help :-) You can't imagine the relief. |
| 19:54.28 | bmoraca_work | vader--: i use Adtran TA900s quite a bit. they're great peices of equipment, but depending on your needs may be a little more expensive |
| 19:54.57 | bmoraca_work | plundra: np |
| 19:55.13 | bmoraca_work | plundra: sometimes it just takes a second set of eyes (or brain) |
| 19:55.16 | vader-- | whats the difference between an IP gateway and a channel bank? |
| 19:55.53 | bmoraca_work | vader--: a channel bank will take a T1 and give you FXS ports...an IP gateway is SIP-to-FXO/FXS |
| 19:56.04 | hardwire | Chainsaw: on top of that.. it won't use TFTP for the logo.bmp |
| 19:56.05 | hardwire | heh |
| 19:56.08 | hardwire | http only plzkthx |
| 19:56.17 | vader-- | i would love to be able to virtualize our asterisk setup |
| 19:56.24 | plundra | bmoraca_work: Yeah, and some sort of reassurance. First time even trying to understand how a PRI might work under the hood :) In combination with all new hardware and software (the dahdi part). |
| 19:56.30 | Chainsaw | hardwire: Indeed, just plain HTTP for that. |
| 19:56.34 | vader-- | we have a decent size virtual cluster with all mission critical support |
| 19:56.35 | plundra | bmoraca_work: It's easy to start doubting your self... |
| 19:56.36 | vader-- | small asterisk server |
| 19:56.39 | p3nguin | Can someone help me with this? http://pastebin.com/d48f08311 Using 1.4.29. |
| 19:56.58 | vader-- | but ive been told virtualizing it is not recommended |
| 19:57.02 | bmoraca_work | vader--: shouldn't be a problem. |
| 19:57.16 | bmoraca_work | vader--: i virtualize asterisk on ESXi all the time...works great |
| 19:57.18 | vader-- | the problem is we have a PRI line and 24FXS ports needed |
| 19:57.25 | vader-- | we don't do voip |
| 19:57.51 | bmoraca_work | vader--: get an Adtran TA924. that'll give you a PRI-to-SIP gateway as well as 24 FXS ports |
| 19:58.10 | vader-- | and it has an ethernet link to the network, plus pri, plus 24fxs? |
| 19:58.26 | *** join/#asterisk ktwilight (n=keliew@123.55-240-81.adsl-dyn.isp.belgacom.be) |
| 19:58.30 | hardwire | Chainsaw: makes me want to spit chickens |
| 19:58.44 | bmoraca_work | vader--: yep |
| 19:58.48 | vader-- | hmm |
| 19:58.59 | bmoraca_work | vader--: great devices, i'm telling you. |
| 19:59.08 | Chainsaw | hardwire: Just wait until you read up on the XML and realise that <40% of what they say actually works on the SIP load. |
| 19:59.37 | bmoraca_work | vader--: or you can use a Cisco router (2801 will do it) to get your PRI to SIP and a separate media gateway for the FXS ports |
| 20:00.09 | vader-- | no clocking issues with vmware? |
| 20:00.21 | bmoraca_work | vader--: i haven't had any, no. |
| 20:00.33 | vader-- | hmm |
| 20:00.43 | bmoraca_work | vader--: that said, my biggest one is 29 phones and i don't do a lot of meetme on them. |
| 20:01.26 | hardwire | Chainsaw: last thought.. any idea how to turn off CDP? |
| 20:01.52 | Chainsaw | hardwire: I don't think you can. |
| 20:02.23 | hardwire | heh |
| 20:02.41 | hardwire | and apparently even thought you can set a TFTP config dir. you can't put dialplan there.. because it only looks at the root. |
| 20:02.45 | hardwire | I'm going to wig out man. |
| 20:02.47 | hardwire | seriously |
| 20:02.54 | hardwire | it's been a while since I had to work with 79xx |
| 20:02.58 | hardwire | and now I remember why |
| 20:03.11 | bmoraca_work | vader--: though that 29 phone one is all g729 and runs an average of 8 concurrent channels...so it does pretty well, I'd say. there are 5 other Asterisk systems and two Windows 2003 servers on that same ESXi box. they all work well. |
| 20:03.32 | vader-- | ya our esx cluster is pretty beefy |
| 20:03.57 | bmoraca_work | vader--: it's one box i paid $400 for :P |
| 20:04.07 | vader-- | im just worried about echoing and stuff |
| 20:04.31 | bmoraca_work | vader--: i have no problems with it and the Adtran has built-in echo cancellation on both the PRI and the FXS |
| 20:05.00 | cusco | mixmonitor() can execute a system command... It executes it, it is a bash script.. but doesn't follow all the steps... I can't understand why! |
| 20:05.04 | cusco | ael has: http://paste.debian.net/57752/ |
| 20:05.09 | bmoraca_work | vader--: though for my virtual PBXes, I have a Cisco AS5400 as my PSTN gateway...but the Adtran will be very similar in performance |
| 20:05.17 | cusco | and /usr/local/bin/mixmon has http://paste.debian.net/57750/ |
| 20:05.30 | cusco | now, it moves the file but doesn't echo it, nor chmod it |
| 20:05.32 | cusco | why? |
| 20:05.43 | vader-- | ya i wish i could find more articles of people using asterisk in esx but they are all from like 2005-2006 and they are like STAY AWAY hehe |
| 20:05.48 | cusco | if I run mixmon by hand in the shell it echoes.. |
| 20:05.51 | vader-- | anyone else in here running asterisk virtual? |
| 20:06.14 | bmoraca_work | don't trust me? :P |
| 20:06.47 | vader-- | i do, just looking for a larger base of people's experiences |
| 20:06.52 | vader-- | :-) |
| 20:08.23 | cusco | ah.. got it |
| 20:08.25 | cusco | I think |
| 20:09.03 | vader-- | i do appreciate the info though |
| 20:09.22 | vader-- | i wish there was some way to get one of these TA924's for testing |
| 20:10.07 | bmoraca_work | vader--: call Adtran and let them know. they sent me one for a few weeks to test out |
| 20:10.13 | bmoraca_work | tell them you might want to resell it |
| 20:11.24 | vader-- | do you know any reps there? |
| 20:13.28 | *** join/#asterisk cmj (n=kathryns@190.241.15.7) |
| 20:14.00 | bmoraca_work | i do, but he's for resellers...i don't know if they would be willing to send one out for just a one-off purchase |
| 20:14.01 | sbrath | to enable BLF in 1.6.2 do I need to set callcounter=yes now instead of call-limit=100 ? |
| 20:14.23 | vader-- | do yuo know what they run? |
| 20:15.48 | shin-shoryuken_ | Anyone know why Asterisk wouldn't pass CNAM data when bridging two PRI channels? |
| 20:16.52 | bmoraca_work | vader--: i can get TA924's for $1178 |
| 20:16.55 | cmj | Does anyone know how to handle a skype user that won't login to SkypeForAsterisk? |
| 20:17.05 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:17.08 | ManxPower-work | sbrath: I think so, but it would be documented in the UPGRADE*.txt files. |
| 20:17.08 | vader-- | what their support like? |
| 20:17.44 | ManxPower-work | shin-shoryuken_: Telcos discard CNAM info for calls from customers. |
| 20:17.53 | shin-shoryuken_ | I'm running a setup with two Asterisk servers, where it's PSTN <= PRI => AstA <= PRI => AstB. |
| 20:18.19 | shin-shoryuken_ | I can see the name show up on AstA, but when the call is passed to the next Asterisk box the name is dropped for some reason. |
| 20:18.21 | ManxPower-work | shin-shoryuken_: that should work |
| 20:18.31 | bmoraca_work | vader--: generally very good. the only issue i ever had to call them about, the tech on the phone had it resolved in 10 minutes |
| 20:18.37 | ManxPower-work | shin-shoryuken_: are you passing the call with a simple Dial? |
| 20:19.32 | p3nguin | Can someone help me with this? http://pastebin.com/d48f08311 Using 1.4.29. |
| 20:19.44 | shin-shoryuken_ | ManxPower-work: Well, it's a Dial command in a Macro - I'm just passing the DAHDI/X and a timeout to it, no extras. |
| 20:20.01 | cmj | I can see my call entering the system, but it is rejected. Due to the problem with the user that wont log in |
| 20:20.10 | voipmonk | is skinny loaded already, p3nguin ? |
| 20:20.42 | ManxPower-work | shin-shoryuken_: as the priority before the dial (not the macro) put something like Noop(CALLERID(all) is ${CALLERID(all)}) and see what the CLID info is just before the dial happens. |
| 20:21.14 | bmoraca_work | vader--: http://tinypic.com/r/taoutt/6 and http://tinypic.com/r/2h3r7h5/6 |
| 20:21.19 | p3nguin | voipmonk: Yes. "The same chan_skinny.so Skinny Client Control Protocol (Skinny) 0" showed up before and after I tried to reload/load it. |
| 20:21.28 | shin-shoryuken_ | ManxPower-work: It's blank - tried it last night. But it shows up correctly when I run 'core show channel DAHDI/X' on the original channel. |
| 20:21.34 | bmoraca_work | vader--: asterisk is pretty reliable on ESX and doesn't take up much resources |
| 20:21.44 | carrar | ttp://tinypic.com/r/2h3r7h5/6 |
| 20:21.46 | carrar | woops |
| 20:21.50 | carrar | damn paste |
| 20:21.59 | ManxPower-work | shin-shoryuken_: it should NOT be blank just before the Dial. If it is then something in your dialplan is screwing it up. |
| 20:22.08 | vader-- | bmoraca i mean if you need to get it replaced will they over night? |
| 20:22.10 | shin-shoryuken_ | ManxPower-work: To clarify, I tried that as the first statement in the receiving context. |
| 20:22.13 | carrar | eww windows |
| 20:22.19 | Katty | did you know that between both of our eyes, and our brains ability to translate it into a video, we see at 324 MP? |
| 20:22.21 | ManxPower-work | shin-shoryuken_: But that is not what I asked you do to. |
| 20:22.44 | shin-shoryuken_ | ManxPower-work: My apologies. I'll run a quick test and see what it comes up with. Thanks! |
| 20:22.54 | Katty | and if you have a 180 degree field of view, that's actually 576 MP |
| 20:22.57 | bmoraca_work | vader--: never had that problem...they carry a 10 year warranty that i believe can include advanced replacement |
| 20:23.16 | ManxPower-work | shin-shoryuken_: if it shows correct then you know the dialplan on your sending server is correct and we can start looking at the PRI settings for DAHDI and Asterisk |
| 20:24.13 | ManxPower-work | I toured The Adtran factory. I was very impressed, and I'm not easily impressed. |
| 20:24.31 | vader-- | ManxPower do you use their stuff? |
| 20:25.49 | carrar | Katty, and how many levels of contrast! |
| 20:26.11 | bmoraca_work | vader--: everything i've seen from them has impressed me...their regional sales director came and met with me personally and they seem to do quite a lot to try and make their customers happy. additionally, their shit is built to last...it's all telco grade |
| 20:27.02 | ManxPower-work | vader--: I even use Adtran for personal stuff. |
| 20:27.23 | shin-shoryuken_ | ManxPower-work: It's coming up blank, using exten => s,2,NoOp(CNAM: ${CALLERID(name)}) |
| 20:27.39 | sbrath | ManxPower-work: I checked the UPGRADE*.txt files, and the answer is unclear |
| 20:27.41 | bmoraca_work | mmm...lunch time...gotta go get some grub...bbiab |
| 20:27.43 | shin-shoryuken_ | ManxPower-work: Sorry, misread. I'll try with 'all'. |
| 20:27.49 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek) |
| 20:28.02 | ManxPower-work | shin-shoryuken_: then it is either not being received from the telco or it's being screwed up some where in your dialplan |
| 20:29.09 | cmj | Does anyone know about skypeforasterisk? |
| 20:30.02 | ManxPower-work | cmj: Digium Sales should, since it's a commercial product |
| 20:30.04 | shin-shoryuken_ | ManxPower-work: Result was NoOp("DAHDI/1-1", "CNAM: "" <my cell number>") |
| 20:30.23 | ManxPower-work | shin-shoryuken_: then it is either not being received from the telco or it's being screwed up some where in your dialplan |
| 20:30.53 | shin-shoryuken_ | ManxPower-work: Could this be a result of setting a custom channel variable? That's all I do, aside from two Goto statements and a Macro call. |
| 20:30.58 | vader-- | so adtran is the recommend voip gateway |
| 20:31.01 | ManxPower-work | shin-shoryuken_: CNAM frequently arrives a fraction of a second after the number. Put a Wait(1) as the first priority for the exten => line that incoming calls match. That would fix it if that's the problem |
| 20:31.20 | cmj | they should, but have been unhelpful. |
| 20:31.27 | ManxPower-work | If it works you should be able to make it something like Wait(.25) or even Wait(.10) |
| 20:33.13 | sbrath | They claim that the call-limit is depricated, and replaced with callcounter.. But I don't really see a dialog on BLF |
| 20:35.01 | [TK]D-Fender | sbrath: And we don't see hint status, configs, call attempts with SIP debug enabled... or much else. |
| 20:35.11 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 20:35.36 | p3nguin | Can someone help me with this? http://pastebin.com/d48f08311 Using 1.4.29. |
| 20:35.39 | sbrath | [TK]D-Fender: Do it's not working for you? |
| 20:35.44 | shin-shoryuken_ | ManxPower-work: Put the Wait in, but no joy. Running the 'core show channel DAHDI/X' command again shows that the 'CallerID Name' section is correct, but the 'CLID' field in the CDR section is just the number. |
| 20:36.23 | ManxPower-work | shin-shoryuken_: Do you have any callerid= lines in your chan_dahdi.conf? |
| 20:36.30 | ManxPower-work | If you do, remove them |
| 20:36.39 | [TK]D-Fender | p3nguin: UNLOAd it. confirm that it is unloaded, then load it |
| 20:37.12 | shin-shoryuken_ | ManxPower-work: They're set to 'asreceived' - I'll remove them. |
| 20:37.55 | ManxPower-work | shin-shoryuken_: asreceived is good. |
| 20:38.08 | ManxPower-work | shin-shoryuken_: pastebin your chan_dahdi.conf |
| 20:38.57 | bmoraca_work | vader--: if i had my way, every gateway i sold would be adtran |
| 20:39.09 | p3nguin | Something is still wrong. http://pastebin.com/d287da0b2 |
| 20:40.41 | hardwire | Chainsaw: neat.. you can't configure speed lines from tftp. |
| 20:41.53 | _abc_ | what does the qualify= option do in conf files? |
| 20:43.11 | Naikrovek | instructs asterisk to continually "ping" the phones (not really a ping, but whatever) and consider the phone offilne if the response is greater than the qualify=parameter |
| 20:43.11 | vader-- | bmoraca are you a reseller, consultant? |
| 20:43.34 | Naikrovek | so if qualify=2000 (the default, I think) and the response from the phone is 2010ms, asterisk will consider the phone to be unreachable |
| 20:43.37 | vader-- | i wonder if i buy one of these on ebay if the warranty will transfer? |
| 20:43.39 | shin-shoryuken_ | ManxPower-work: http://pastebin.com/mf3e8d |
| 20:43.41 | ManxPower-work | _abc_: It sends a SIP OPTIONS packet and measures the response time. It is also used to keep a connection "chatty" so NAT translations don't timeout. |
| 20:44.07 | bmoraca_work | vader--: i am a reseller, yes. i don't know whether or not the warranty transfers. |
| 20:45.18 | shin-shoryuken_ | ManxPower-work: SERVER2 was originally 'switchtype=national' - ni1 was a hail mary I tried last night. Same results either way. |
| 20:45.59 | *** join/#asterisk hfb (n=hfb@98.112.226.53) |
| 20:46.10 | ManxPower-work | shin-shoryuken_: Try this: http://pastebin.com/m28d9f02 |
| 20:46.39 | ManxPower-work | see if that makes the callerid show in that noop you have |
| 20:47.02 | vader-- | bmoraca do you have a 924 running PRI and FXS at the same time? |
| 20:47.08 | ManxPower-work | also put the callerid=asreceived back in as well |
| 20:47.24 | bmoraca_work | vader--: not a 924, no, but I have 904s and 908es running both at the same time |
| 20:47.26 | shin-shoryuken_ | ManxPower-work: Will do, thanks. Any settings in there need a full reload, or will 'reload chan_dahdi.so' suffice? |
| 20:47.54 | ManxPower-work | shin-shoryuken_: I don't know. I would to a module unload chan_dahdi.so then module load chan_dahdi.so |
| 20:47.59 | bmoraca_work | vader--: i typically use them in the reverse (SIP to PSTN converted to PRI for legacy PBX), but there's no reason why they shouldn't work |
| 20:48.35 | ManxPower-work | shin-shoryuken_: I assume calls are coming in on "PSTN PRI" |
| 20:48.58 | *** join/#asterisk keith (i=meshuga@adsl-69-87-6-143.dsl.irvnca.interirc.net) |
| 20:49.30 | shin-shoryuken_ | ManxPower-work: Correct. |
| 20:49.36 | keith | anyone ever used SIPP as a UAS w/ authentication? |
| 20:49.46 | *** join/#asterisk slidesinger (n=slidesin@c-68-44-99-50.hsd1.nj.comcast.net) |
| 20:50.07 | *** join/#asterisk ajohnson (n=aaron@65-122-4-130.dia.static.qwest.net) |
| 20:52.24 | bmoraca_work | vader--: unfortunately, i don't have my lab set up right now so I can't verify 100%, but by the way they work, I am confident in recommending that as a solution for you |
| 20:53.23 | ManxPower-work | There is also a huge 2nd hand market for Adtran stuff. More so for their older stuff than their newer stuff. |
| 20:55.37 | bmoraca_work | TA600s aren't much good for much, but their Atlas 550s are pretty freakin awesome for lots of stuff |
| 20:55.49 | vader-- | i called adtran the presales design guy doesn't think the 924 can do both pri and fxs traffic at the same time |
| 20:56.17 | bmoraca_work | vader--: it can, believe me |
| 20:56.35 | ManxPower-work | vader--: if it has the interfaces then it should be able to do it. obviouslly you can't run PRI and FXS on the same channels. |
| 20:56.45 | *** join/#asterisk rycar (n=rycar@watsontouchstone.com) |
| 20:56.58 | bmoraca_work | vader--: telcos use these to provide "integrated T1s" and analog lines to people. i use them for that same purpose, and i can verify they work |
| 20:57.07 | *** join/#asterisk codefreeze-lap (n=murf@mail.parsetree.com) |
| 20:57.11 | ManxPower-work | That's what we use them for. |
| 20:57.18 | ManxPower-work | Well, one of the things, at least. |
| 20:58.15 | rycar | so, one of my asterisk boxes got hacked. =) |
| 20:58.41 | *** join/#asterisk fogrider (n=fog@209.248.173.89.nw.nuvox.net) |
| 20:59.19 | bmoraca_work | ManxPower-work: what else do you use them for? |
| 20:59.28 | fogrider | I got a polycom 501 IP phone, but i don't have any ac adapter. phone doesn't work PoE i guess without an adapter? is this true? anybody got any experience with these phones? |
| 20:59.56 | Naikrovek | poe does not require an adapter |
| 21:00.01 | bmoraca_work | rycar: i had one get compromised as well...or, rather, they exploited a hole in phpMyAdmin that allowed them to upload a paypal spoof page...only lasted about a day though |
| 21:00.07 | bmoraca_work | Naikrovek: on a 501 it does |
| 21:00.11 | Naikrovek | really |
| 21:00.13 | Naikrovek | wow |
| 21:00.13 | bmoraca_work | yep |
| 21:00.14 | Naikrovek | okay |
| 21:00.19 | Naikrovek | death to 501 |
| 21:00.34 | [TK]D-Fender | Naikrovek: was disco'd some time ago |
| 21:00.40 | Naikrovek | thankfully |
| 21:00.41 | rycar | this one they have an autodialer calling everyone in the 301 area code telling them that their card has been deactivate and to enter their pin number to turn it back on |
| 21:01.05 | fogrider | bmoraca_work: any way of making a home-made one? I do cabling so I can inject my own power just not sure what power requirements are since it doesn't follow the 48v standard. |
| 21:01.17 | rycar | I had to null route 64.34.179.89 to get it to stop |
| 21:01.23 | fogrider | bmoraca_work: any Ideas? or do i need to buy some stupid adapters? |
| 21:02.14 | bmoraca_work | fogrider: i have no idea, but this is the calbe that's needed: http://www.voiplink.com/Polycom_PoE_Cable_for_501_and_301_IP_Phones_p/polycom-poe-cable-cisco.htm |
| 21:02.25 | bmoraca_work | fogrider: might be able to find out how to build one on some other sites |
| 21:02.54 | [TK]D-Fender | nope |
| 21:03.14 | [TK]D-Fender | Polycom's cable has PoE negotiation circuitry in it |
| 21:03.22 | [TK]D-Fender | this is not a question of mere pinout |
| 21:03.23 | fogrider | bmoraca_work: well i looked at a few sites and didn't get much luck, thought I would see if I could find somebody that had some experience with this model. thanks for the link. |
| 21:03.28 | bmoraca_work | i didn't think so |
| 21:03.44 | vader-- | i.e. if a call comes in on the pri and is designated for fxs he said it might not work |
| 21:03.53 | fogrider | D-Fender: ok. question. do i still need a PoE switch if I get the cable? |
| 21:04.24 | vader-- | he also said that the 924 won't take PRI from the telco only from PBX side, and that i would need the 924e to get PRI working from the telco |
| 21:04.25 | bmoraca_work | vader--: no, it probably won't. it's not meant to be switched like that. you'd have to send it to asterisk and then back to the Adtran. it has to do with the way the FXS ports are |
| 21:04.29 | fogrider | D-Fender: would rather get just adapters for the phones since I don't have a PoE switch at the location of the phones, so don't wanna have to buy two things |
| 21:04.36 | vader-- | :-/ |
| 21:04.37 | *** join/#asterisk codefreeze-lap (n=murf@mail.parsetree.com) |
| 21:04.45 | bmoraca_work | vader--: that's not true, you just have to use a T1 crossover instead most likely |
| 21:06.18 | [TK]D-Fender | [16:03]<fogrider>D-Fender: ok. question. do i still need a PoE switch if I get the cable? <= of course |
| 21:06.18 | [TK]D-Fender | fogrider: thats the point |
| 21:06.18 | [TK]D-Fender | fogrider: get the PoE cable so you don't need a WALL WART |
| 21:06.18 | fogrider | D-Fender: ok, another question then, is there a way to power the 501 with PoE? |
| 21:06.21 | [TK]D-Fender | fogrider: otherwise the phones typically come with a similar cable that the wall-wart plugs into in-line |
| 21:06.38 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 21:07.12 | bmoraca_work | fogrider: that cable i linked for you IS the cable required to do PoE. there's a different cable for plugging in to an AC outlet |
| 21:07.13 | [TK]D-Fender | fogrider>D-Fender: ok, another question then, is there a way to power the 501 with PoE? <- you seem to have trouble following. There are *2* special cables to power a 501. One is a PoW cable with dongle in-line. Second is one that you plug a wall-wart into. |
| 21:07.35 | [TK]D-Fender | fogrider: the phone should come with ONE of these 2 |
| 21:08.16 | [TK]D-Fender | fogrider: don't get the PoE cable only to use a single-port injector right alongside it. |
| 21:09.00 | fogrider | D-Fender: thanks for explaining... phones were purchased by boss off ebay. guess what? no power adapters at all so i was a little confused :) |
| 21:10.05 | *** join/#asterisk tacvbo (n=tacvbo@189.146.173.71) |
| 21:10.17 | bmoraca_work | vader--: a T1 is a T1...as long as you match up your framing/coding, get your timing right, and get your dialplan correctly done, it'll connect. there is no software difference between the 924 and the 924e, so there's no reason that anything in the 924e wouldn't work on a 924...the only difference is that the channelized port in the 924 is already crossed, meaning you'd need to cross it again to plug it in to the telco |
| 21:10.45 | [TK]D-Fender | fogrider: they should have come with the wall-wart cable at least |
| 21:12.46 | fogrider | D-Fender: they didn't. I will have the boss see if they were suppose too. so just so I understand, this wall-wart cable your taking about involves a power adpater that plugs into the wall outlet at the phone location and then pulls into this adapter which then plugs into the phone? so do i need a transformer and the adapter or is it just one piece? just wanna make sure i get the right stuff when i buy it |
| 21:13.13 | Katty | heh. sony person is retarded. they cannot tell me if the model number i am looking at supports usb streaming or not. |
| 21:13.19 | Katty | in fact, they don't even know what usb streaming is. |
| 21:13.28 | Katty | derrrr it has a usb port |
| 21:13.44 | Katty | well of course it has a usb port, but that's not what i'm asking! >.< |
| 21:14.20 | [TK]D-Fender | fogrider: picture an ethernet wire with a nugget in the middle. the wall wart plugs into the nugget like a "T" |
| 21:15.09 | fogrider | D-Fender: ok, i didn't get the ethernet with nuggert or the wall wart. ok, i will go from there. thanks again. |
| 21:16.51 | *** join/#asterisk Deeewayne (n=dwayne@adsl-070-148-053-233.sip.bhm.bellsouth.net) |
| 21:16.51 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 21:17.24 | fogrider | I got another question involving g729a. I got my asterisk passing-thru g729a from my sip provider to a sip phone but my understanding is to create the stream i need licsenses per channel. I have two iphones that i would like to always use g729a so i guess i need to channnel liscenses. i see a few provides (howler and digium) who provide them. Who is the best and easy to install? the computer has 1.4 asterisk and was installed using asterisk |
| 21:17.55 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 21:17.57 | ManxPower-work | fogrider: get the one from digium |
| 21:18.03 | _abc_ | Katty: what is 'usb streaming' ? |
| 21:18.10 | ManxPower-work | never heard of "howler" |
| 21:18.14 | [TK]D-Fender | fogrider: You only need licenses the moment * has to transcode anything to them. This includes ringing sounds generated by * |
| 21:19.16 | _abc_ | using a licensed codec with an open source pbx defeats the purpose |
| 21:19.37 | [TK]D-Fender | fogrider: If every recording sound, and bridged call uses only G.729 then you're OK without licenses. Meetme is an exception as ALL calls get transcoded |
| 21:19.49 | [TK]D-Fender | _abc_: Not in the slightest |
| 21:20.26 | [TK]D-Fender | _abc_: I'm using an Open Sourse Operating System on a PROPRIETARY LICENCED CPU> OH NOES! |
| 21:20.58 | _abc_ | [TK]D-Fender: rip it out quick! |
| 21:23.15 | _abc_ | how does the unistim qualify= option work? it seems to use a ridiculously long timeout |
| 21:23.37 | _abc_ | i set the qualifyfreq=10 but it times out after ages |
| 21:23.39 | Katty | _abc_: using it like a webcam |
| 21:23.46 | [TK]D-Fender | _abc_: You're using UNISTIM PHONES and whining that G.729 is proprietary? |
| 21:23.49 | [TK]D-Fender | HERESY |
| 21:23.54 | Katty | _abc_: sony's 'catch phrase' for it is called USB Streaming |
| 21:23.58 | _abc_ | [TK]D-Fender: the phones are not at my end |
| 21:24.06 | Zhad | :-) |
| 21:24.12 | [TK]D-Fender | _abc_: No absolution for you! |
| 21:24.18 | Zhad | is unistim the nortel one? |
| 21:24.26 | [TK]D-Fender | Zhad: Yup |
| 21:24.47 | Zhad | Thought do |
| 21:25.23 | _abc_ | Katty: oh, you know, there is no streaming per se. streaming consists of successive individual images shot at the frame frequency of which the camera is capable |
| 21:25.54 | _abc_ | and this is one of the reasons for which firewire flies hoops around usb and will never be replaced in the av segment |
| 21:26.32 | Zhad | takes it that no-one here has ever had a Zultys WIP2 ? |
| 21:26.41 | [TK]D-Fender | ~wifivoip |
| 21:26.42 | infobot | [~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended. Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc |
| 21:27.04 | Zhad | :-) |
| 21:27.26 | Zhad | admittedly, 3 out of 4 of the wireless phones I've used have been pretty poor. |
| 21:27.35 | _abc_ | s**t more snow and it's -12 C outside now |
| 21:27.43 | [TK]D-Fender | Zhad: the 4th is due to show its true colours momentarily |
| 21:27.46 | Zhad | and since a firmware update, the voip part of my e71 doesn't work properly. |
| 21:27.50 | [TK]D-Fender | Alrighty, checkout time, later all. |
| 21:28.03 | _abc_ | l8 |
| 21:28.12 | Zhad | thinks he meant to type 4 out of 4. |
| 21:28.30 | _abc_ | Zhad: wireless phones are like ink cartridges. meant to last a very short time |
| 21:28.45 | _abc_ | omg what a winter |
| 21:29.02 | _abc_ | this year canada is balmy compared to europe |
| 21:29.20 | bmoraca_work | _abc_: it's been raining for 8 days straight in Sunny California |
| 21:29.26 | Zhad | I've had my e71 for a year, and it only started playing up (sound stutters) since a firmware update I did a couple of weeks ago. |
| 21:29.45 | Katty | _abc_: oh i /will/ find a camcorder that supports streaming (= |
| 21:29.53 | bmoraca_work | Zhad: that's why you don't try to fix what's not broke :) |
| 21:30.05 | rycar | sigh, going to have to reinstall this whole server. damn hackers |
| 21:30.31 | Zhad | bmoraca> the update was supposed to make the useless browser more stable. (it didn't). |
| 21:30.46 | _abc_ | bmoraca_work: well at least there are no bush fires consuming mansions. |
| 21:30.50 | *** join/#asterisk pawz (n=pawz@210.56.86.70) |
| 21:31.16 | bmoraca_work | rycar: why? fix whatever hole they got in through and use hosts.allow and hosts.deny to restrict who can access what services |
| 21:31.42 | _abc_ | rycar: you are right if you were hacked reinstall the server |
| 21:31.45 | ManxPower-work | rycar: secure your server next time. |
| 21:31.50 | _abc_ | and that |
| 21:31.55 | ManxPower-work | Are you sure they are not just using a hole in your dialplan? |
| 21:32.05 | batphone | lulz |
| 21:32.06 | rycar | they were connecting over IP |
| 21:32.16 | _abc_ | what kind of hole can you have in the dialplan to hack a server?! without agi and so on |
| 21:32.20 | ManxPower-work | like you forgetting to put allowguest=no in sip.conf or not making sure the context= line in [default] points to a restricted context? |
| 21:32.39 | Zhad | bmoraca> Fail2ban is your friend. |
| 21:32.45 | batphone | is asterisk good for CPS |
| 21:32.52 | batphone | like 200 CPS |
| 21:32.57 | ManxPower-work | rycar: SSH over IP or SIP over IP? |
| 21:33.18 | Zhad | also permit= lines in your user declarations is handy. |
| 21:33.22 | rycar | I'm pretty sure it's SIP, can't find any evidence of SSH being used |
| 21:33.28 | ManxPower-work | rycar: I suspect you simply had an unsecure dialplan. |
| 21:33.37 | *** join/#asterisk RobH (n=robh@rob.tech.wikimedia.org) |
| 21:33.45 | ManxPower-work | rycar: DO you have allowguest=no in your sip.conf? |
| 21:33.47 | rycar | also the server version is Asterisk 1.2.14 svn rev 48468 |
| 21:33.51 | *** join/#asterisk bbkt-trix (n=bbkt-tri@unaffiliated/bbkt-trix) |
| 21:33.58 | _abc_ | how do you define 'hacked' ?! they hacked the unix server or just made free calls ? |
| 21:34.10 | Zhad | And default context one that is safe. |
| 21:34.22 | dlynes_laptop | rycar, why are you using a subversion version of asterisk...and one so old at that? |
| 21:34.28 | _abc_ | how do you define 'hacked' ?! they hacked the unix server or just made free calls ? |
| 21:34.33 | _abc_ | rycar: |
| 21:34.40 | rycar | it's a place I used to work at a long time ago, they called me when the server went nuts |
| 21:34.42 | ManxPower-work | _abc_: sounds like "hacked=walked thru open door" |
| 21:34.52 | *** join/#asterisk bbkt-trix (n=bbkt-tri@unaffiliated/bbkt-trix) |
| 21:34.52 | ManxPower-work | rycar: you have not answered my question. |
| 21:34.52 | _abc_ | ManxPower-work: to where?! |
| 21:35.15 | Katty | Panasonic SDR-H80K apparently supports it :> |
| 21:35.21 | rycar | I don't see an allowguest entry in my sip.conf |
| 21:35.24 | ManxPower-work | rycar: Do you or do you not have allowguest=no in the [general] section of sip.conf? |
| 21:35.24 | Zhad | He might want to allow guests, for e164s sake :-p |
| 21:35.24 | Katty | checks user manual to make sure |
| 21:35.48 | ManxPower-work | rycar: if you do not have allowguest=no then users will be able to connect into the server with no auth at all. |
| 21:36.09 | ManxPower-work | i.e. allowguest defaults to "yes" |
| 21:36.09 | Zhad | Isn't it only in the default context though? |
| 21:36.17 | _abc_ | i think that * fails where it tries to be too many things at once |
| 21:36.23 | ManxPower-work | Zhad: we don't know what his default context is yet. |
| 21:36.28 | _abc_ | like auth server and e164 broker |
| 21:36.32 | rycar | all the calls in the log show up as being made from one of the local SIP phones, I wonder if they guessed the password to that extension |
| 21:36.49 | dlynes_laptop | rycar, are your sip passwords easily guessable? |
| 21:36.50 | Zhad | althouth iirc there was a security advisory a while back iirc, regarding fake authentications. |
| 21:36.53 | _abc_ | rycar: how local is the phone? |
| 21:36.54 | Zhad | was a long time ago though. |
| 21:37.08 | rycar | local as in same LAN as the server |
| 21:37.08 | ManxPower-work | rycar: no. Asterisk will happily accept any username the caller provides unless you have allowguest=no |
| 21:37.09 | _abc_ | rycar: can someone get to it through the hetwork and has that phone a known exploit? |
| 21:37.10 | dlynes_laptop | rycar, did you assign the extension number to the sip user/peername? |
| 21:37.26 | Zhad | _abc_> nah, it's only failing is that it can use t'internet :-p |
| 21:38.05 | ManxPower-work | rycar: does the sip account for that phone have a bad password? |
| 21:38.08 | _abc_ | Zhad: not true, but trying to be too many things at once results in not being good at anything in particular |
| 21:38.22 | ManxPower-work | _abc_: Asterisk is fine, the user is the issue. |
| 21:38.39 | ManxPower-work | I feel that allowguest should default to "no", however. |
| 21:39.08 | Zhad | maybe in 1.8 |
| 21:39.22 | Zhad | sorry, 1.6.3 |
| 21:39.29 | Zhad | is tired. |
| 21:39.53 | Zhad | I get annoyed when default behaviour changes in a minor release |
| 21:39.56 | dlynes_laptop | it matters not...what matters is that someone defined a default context |
| 21:40.11 | dlynes_laptop | and that's why he's got a whole bunch of free calls |
| 21:40.17 | ManxPower-work | dlynes_laptop: yes, but explaining and solving that is much harder than adding allowguest=no |
| 21:40.22 | _abc_ | ManxPower-work: PBHP ? (problem between handset and phone) |
| 21:40.23 | rycar | SIP ports were open on the router so that the users could take their phones home and work while on the road |
| 21:40.36 | rycar | weird thing is that even when I closed the ports, the calls kept going out |
| 21:40.47 | Zhad | _abc_> If you don't have it, and your default context can make outgoing calls, then it's not even a hack to make calls though it, it's juck a configured handset. |
| 21:40.48 | ManxPower-work | rycar: well stop leaving the door to the PBX open with allowguest=yes |
| 21:41.19 | bmoraca_work | allowguest=yes is soooo much fun, though! |
| 21:41.20 | dlynes_laptop | ManxPower-work, i'm guessing that option's been around since before asterisk 1.0? |
| 21:41.33 | Zhad | Also, stop using numbers as names for SIP users |
| 21:41.46 | ManxPower-work | You have your answer, I'm not going to try to convince you that it is the right answer. |
| 21:41.56 | Zhad | get something like Fail2Ban to automatically cut off anyone who makes a lot of successive auth failures |
| 21:41.57 | ManxPower-work | dlynes_laptop: no, I think it's new in 1.2 or 1.4. |
| 21:42.03 | dlynes_laptop | and don't name your main dialplan context 'default' |
| 21:42.23 | ManxPower-work | dlynes_laptop: I always set my default context to INVALID |
| 21:42.29 | dlynes_laptop | that's gotta be the stupidest thing i've ever seen |
| 21:43.02 | ManxPower-work | dlynes_laptop: it worked since before there was allowguest=no |
| 21:43.12 | Zhad | use permit= on user declarations to tie them to either the ip they'r eusing at work, or that and the block they're using at home. |
| 21:43.17 | rycar | I don't have an 'allowguest' in any of my conf files. I wish I knew about that before |
| 21:43.27 | ManxPower-work | rycar: put it in right now. |
| 21:43.34 | dlynes_laptop | ManxPower-work, exactly |
| 21:43.38 | ManxPower-work | reload SIP, unblock the IP and see what happens. |
| 21:43.46 | Zhad | upgrade to a more recent asterisk, the amount of work that's been done on the sip channel driver since then is astounding. |
| 21:43.51 | ManxPower-work | make sure the "phone" that was making the calls doesn't have a stupid password |
| 21:44.19 | dlynes_laptop | ManxPower-work, I've had to fix other people's code that did similar stupid things...inbound and outbound were both running in the same context, and it was appropriately named 'default', which coincidentally was also named as the default context in the sip general section |
| 21:44.32 | Zhad | eww |
| 21:44.32 | dlynes_laptop | talk about moronic |
| 21:44.46 | dlynes_laptop | and that was set up by someone that should have known better...he was just too lazy to do it properly |
| 21:45.07 | Zhad | have they set up any more since? |
| 21:45.28 | dlynes_laptop | Zhad, many...he's the ceo of a couple of voip companies |
| 21:45.34 | praet | are freepbx questions ok in here? the main chan seems to be sleeping |
| 21:45.55 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:45.57 | Kobaz | praet: you can ask, but most likly noone's gonna know the answer |
| 21:45.59 | ManxPower-work | praet: no. |
| 21:46.06 | keith | anyone ever used SIPP as a UAS w/ authentication? |
| 21:46.29 | Zhad | hasn't |
| 21:46.29 | dlynes_laptop | praet, just ask in there, even if they seem to be sleeping |
| 21:47.10 | dlynes_laptop | praet, it might take a couple of hours before it gets answered, but it'll generally eventually get answered as long as it's not in an xxxx_custom.conf file |
| 21:47.34 | Nugget | eyes [TK]D-Fender |
| 21:47.37 | ManxPower-work | dlynes_laptop: so pretty much he'll get an answer on the #FreePBX channel anyway. |
| 21:47.59 | praet | k :) |
| 21:48.04 | ManxPower-work | If it was up to me all FreePBX users would be kick/banned from #Asterisk. Be glad it's not up to me. |
| 21:48.06 | dlynes_laptop | more than likely....almost zero questions on that channel about the custom configs |
| 21:48.24 | Nugget | ManxPower-work: common cant you just tell me what to type to make it work?!?! kk |
| 21:48.34 | dlynes_laptop | telnet |
| 21:48.34 | Nugget | telnet is eeeeeeevil! |
| 21:48.43 | Zhad | shutdown -h now |
| 21:48.52 | Kobaz | dd if=/dev/zero of=/dev/sda |
| 21:49.01 | praet | easy guys... |
| 21:49.03 | Zhad | dd if=/dev/zero of=/dev/sda |
| 21:49.05 | *** join/#asterisk muiro (n=muiro@unaffiliated/muiro) |
| 21:49.10 | Zhad | dammit |
| 21:49.14 | Kobaz | beat you! |
| 21:49.26 | Naikrovek | nugget isn't stupid |
| 21:49.36 | Zhad | that's only because I was stumbling over remembering the switch for blocksize |
| 21:49.38 | muiro | in MeetMe, is there any way I can set a user to talk only mode /after/ the conference has started? |
| 21:49.40 | Nugget | at least not about that. :) |
| 21:49.43 | Naikrovek | his thing is limited to telnet and certain kinds of unmentionable material |
| 21:50.13 | dlynes_laptop | Naikrovek, bras and panties? |
| 21:50.26 | Naikrovek | that is mentionable :) |
| 21:50.38 | Naikrovek | nevermind, i don't wanna take the joke that far |
| 21:50.43 | dlynes_laptop | well...they're usually referred to as a lady's unmentionables :0 |
| 21:50.46 | [TK]D-Fender | Telnet is not evil. It is a wide open door witha sign saying "Evil welcome, come on in" :p |
| 21:50.55 | Nugget | heh |
| 21:50.56 | Zhad | telnet is very useful |
| 21:51.00 | Naikrovek | yes telnet is just an unsecured conduit |
| 21:51.08 | Zhad | for testing stuff like mail servers and web servers. |
| 21:51.16 | Naikrovek | he's not talking about that kind of telnet |
| 21:51.20 | Nugget | whatever you do, just don't telnet to me. |
| 21:51.29 | Naikrovek | he's talking about administering remote machines over the internet via telnet |
| 21:51.31 | Zhad | or connecting to talkers |
| 21:51.32 | Naikrovek | which is how it used to be done |
| 21:51.42 | dlynes_laptop | Nugget, i wouldn't dream about it....that's what dcc ctcp is for |
| 21:51.44 | Zhad | remembers |
| 21:52.01 | dlynes_laptop | Zhad, omg...evil memories |
| 21:52.08 | rycar | ssh is the new telnet |
| 21:52.11 | dlynes_laptop | Zhad, obviously you're from the pacific rim |
| 21:52.17 | Zhad | talk dlynes |
| 21:52.26 | Naikrovek | when i worked at verio there wasn't a firewall anywhere on our network. i was in kansas, and i would administer routers in IL, SD, TX, OK, and a few others all via telnet |
| 21:52.30 | rycar | I remember the good ol days when I allowed passwords on sshd |
| 21:52.35 | Zhad | is from/in england |
| 21:52.49 | dlynes_laptop | Zhad, oh....talkers were quite common with pacific islanders |
| 21:53.03 | dlynes_laptop | Zhad, more popular even than mud |
| 21:53.04 | Naikrovek | wishes his phone would just upload its logs already geez |
| 21:53.09 | Zhad | is on one atm. |
| 21:53.30 | dlynes_laptop | ah...wow...didn't think they were still used |
| 21:53.33 | Zhad | couldn't understand the appeal of muds. |
| 21:53.37 | Naikrovek | what's a talker |
| 21:53.37 | dlynes_laptop | I never did see much point to talkers |
| 21:53.51 | dlynes_laptop | Naikrovek, it's like a mud, but it's purely for chatting...no game play |
| 21:53.57 | Naikrovek | like irc |
| 21:53.58 | rycar | muds were great. I played on a star trek one back in the day that is better than most MMOs today |
| 21:54.04 | *** part/#asterisk muiro (n=muiro@unaffiliated/muiro) |
| 21:54.04 | Zhad | someone I know, wrote this one to keep in touch with his dad while he was at university |
| 21:54.25 | Kobaz | there was that one weird little network for chatting, luna or something |
| 21:54.28 | Zhad | 18/19 years ago. |
| 21:54.34 | rycar | you could fly the ship around in space, ships took multiple crewmembers to control, the holodeck was user programmable |
| 21:54.36 | Kobaz | there was maybe like 10 channels and 50 people |
| 21:55.00 | dlynes_laptop | there's also mushes, mucks, moos, ... |
| 21:55.11 | Kobaz | i played muds in the 90's |
| 21:55.14 | dlynes_laptop | all huge time wasters |
| 21:55.14 | *** join/#asterisk adam0563 (n=adam0563@173-14-238-70-Utah.hfc.comcastbusiness.net) |
| 21:55.20 | dlynes_laptop | even more so than irc |
| 21:55.25 | Kobaz | heh |
| 21:55.36 | batphone | eats a grue |
| 21:55.46 | Zhad | at least you weren't of the age where you used terminal software to connect to them then |
| 21:56.30 | Kobaz | i never played a mud on a bbs, but i've been on plenty of dialup bbs's |
| 21:56.47 | jaytee | looks back fondly at the days of perforated paper tapes |
| 21:57.03 | Kobaz | is thankful those days are gone |
| 21:57.07 | Zhad | I remember seeing advertised a game that was literally a dial-in mud, without being sat on a bbs. |
| 21:57.43 | Kobaz | there's still some majorbbs servers up, listening on telnet, running majormud |
| 21:57.44 | [TK]D-Fender | was a TradeWars2002 God... |
| 21:57.51 | Zhad | ATDT 09061234567 |
| 21:57.53 | Zhad | go north |
| 21:58.29 | Zhad | a few years ago I saw one running in a java applet |
| 21:58.37 | Zhad | dunno if that's still going. |
| 21:58.44 | [TK]D-Fender | pummels Zhad with a stack of Zork floppies |
| 21:58.52 | rycar | wow, some idiots actually entered their card and pin numbers |
| 21:58.53 | eppigy | 8[] |
| 21:59.07 | Zhad | rycar> ? |
| 21:59.15 | adam0563 | alright so I finally got incoming phone calls coming into my PBX, however when I make outbound calls it doesn't work. http://pastebin.com/m288790a7 <- that is the part of the dialplan and it says http://pastebin.com/m9752f3d in the Asterisk CLI |
| 21:59.22 | Kobaz | rycar: ask and ye shall recieve |
| 21:59.39 | rycar | whoever was using my system was autodialing people and a recorded voice was asking for card and pin numbers |
| 21:59.45 | voipmonk | adam0563: show some debug |
| 21:59.55 | rycar | I had call recording on all calls turned on |
| 21:59.57 | [TK]D-Fender | adam0563: Show your DAHDI CONFIGS |
| 22:00.00 | Kobaz | adam0563: your card isn't configured right |
| 22:00.09 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 22:00.27 | Zhad | isn't the u<maiulbox> deprecated now? |
| 22:00.31 | Kobaz | yes |
| 22:00.45 | Zhad | iirc u and b are now switches |
| 22:00.48 | rycar | well, gotta run, thanks for all the help |
| 22:01.22 | Zhad | wonders if rycar is about to get prosecuted for fraud. |
| 22:02.12 | Zhad | You honour, the defense is quite simple. I shall prove to you that my client wasn't competent enough to commit this level of fraud. |
| 22:02.31 | Kobaz | heh |
| 22:02.46 | *** join/#asterisk daidoji (n=daidoji7@99.48.50.198) |
| 22:05.33 | Zhad | Am I the only one that doesn't use numbers for sip users/peers? |
| 22:05.40 | Nugget | I don't use numbers. |
| 22:05.59 | ManxPower-work | Zhad: I use the MAC with -a, -b, -c, etc appended to it for the line appearance |
| 22:06.35 | Nugget | ManxPower-work: doesn't that get annoying with the inevitable phone swapping and replacement that takes place over time? |
| 22:06.57 | ManxPower-work | Nugget: no. |
| 22:07.48 | ManxPower-work | when a call hits an extension a set of variables describing the destinations is set then a macro takes over from there to route the call based on that. |
| 22:08.09 | Kobaz | it's all just labels |
| 22:08.18 | Nugget | *nod* |
| 22:08.21 | Nugget | that makes sense |
| 22:08.22 | Kobaz | there's no technical difference between sip/1234 and sip/bob |
| 22:08.23 | ManxPower-work | someone gets a new phone, update the server with the new MAC |
| 22:09.12 | ManxPower-work | Since many times one extension might ring on more than one phone, it's handy to ask for the MAC from the bottom of the phone to know for sure that want they are saying actually jives with reality. |
| 22:09.17 | Nugget | I have a script that autogenerates my phone configs on the tftp server based on mac from templates. so I'm sort of doing the same thing, just at a lower level. |
| 22:09.34 | Nugget | a master config file that maps mac address to sip username base |
| 22:09.55 | Nugget | the dialplan is the same, but the phones get mapped as changes occur |
| 22:10.18 | ManxPower-work | Nugget: at my last job we might have had one add/move/change a week normally |
| 22:10.48 | *** join/#asterisk xpot-mobile (n=xpot@173-14-232-121-Utah.hfc.comcastbusiness.net) |
| 22:11.12 | *** join/#asterisk anthm][ (n=anthm@CPE-72-128-89-126.wi.res.rr.com) |
| 22:16.23 | dlynes_laptop | speaking of autoconfigs |
| 22:16.44 | dlynes_laptop | Are there any voip devices that you can actually always autoconfig? |
| 22:17.28 | dlynes_laptop | Every device I've run across seems to have issues every once in a while with autoconfigging from tftp |
| 22:17.37 | dlynes_laptop | and when they start having problems, they never stop having problems |
| 22:17.38 | *** join/#asterisk TimeRider (n=steve@78.32.26.1) |
| 22:17.49 | Kobaz | polycoms have no problem always doing an autoconfig |
| 22:17.53 | ManxPower-work | dlynes_laptop: the only time I've had polycoms fail to FTP their configs is when I screwed something up on the server. |
| 22:18.18 | Kobaz | the moral of the story: don't screw up the server |
| 22:18.42 | ManxPower-work | I imagine that TFTP might could fail in many kinds of packet loss situations. |
| 22:18.59 | ManxPower-work | TFTP's "error recovery" is to retransmit the packet up to three times and then give up. |
| 22:19.16 | Kobaz | i always provision over ftp, it's much more straightforward, and it's polycom's default, so i don't have to change anything on the phone |
| 22:19.58 | Kobaz | if you needed to provision over the internet, ie: make it through nat... you should use https |
| 22:20.51 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:21.47 | ManxPower-work | How would the phone upload any config changes using https? |
| 22:22.06 | Kobaz | you wouldn't |
| 22:22.16 | Kobaz | unless the phone supported dav |
| 22:22.28 | ManxPower-work | so if you use https provisioning the phone can't upload any configs, logs, etc? |
| 22:22.40 | Kobaz | i don't think so... never tried it though |
| 22:22.55 | ManxPower-work | I'll have to read that section of the admin guide |
| 22:23.04 | Kobaz | polycom might support dav, who knows... it's got the kitchen sink |
| 22:24.25 | *** join/#asterisk aidinb (n=Aidin@h-67-103-140-66.lsanca54.static.covad.net) |
| 22:27.07 | *** join/#asterisk salparadise (n=sal@c-67-172-255-70.hsd1.ut.comcast.net) |
| 22:29.26 | *** join/#asterisk rossand (n=aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 22:37.51 | adam0563 | here is my debug info: http://pastebin.ca/1767171 |
| 22:38.11 | *** join/#asterisk bcrisp (n=bcrisp@70.102.242.138) |
| 22:38.20 | bcrisp | hey all |
| 22:38.30 | *** join/#asterisk RobH (n=robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 22:41.27 | bcrisp | Getting an "Unable to write frame" error when using Dial(): http://pastebin.ca/1767217 |
| 22:42.58 | dlynes_laptop | ManxPower-work, Kobaz thanks for the info |
| 22:45.01 | bcrisp | the issue only shows up when the number dialed is a cell # (so far) |
| 22:46.08 | *** join/#asterisk hluesea (n=hulusika@88.247.127.66) |
| 22:46.37 | *** join/#asterisk adam0563 (n=adam0563@173-14-238-70-Utah.hfc.comcastbusiness.net) |
| 22:46.47 | Zhad | had that before |
| 22:46.52 | Zhad | can't remember what it is |
| 22:47.17 | bcrisp | it appears to go directly to voicemail on the cell |
| 22:47.43 | ManxPower-work | bcrisp: usually that means "caller hung up" |
| 22:47.45 | ManxPower-work | [Jan 26 15:35:06] WARNING[29958]: app_dial.c:807 wait_for_answer: Unable to write frame |
| 22:48.08 | bcrisp | i've tested it with my own cell #, office IP phone etc |
| 22:48.16 | bcrisp | cell # doesnt ring but a VM shows up |
| 22:48.18 | Zhad | don't Grandstreams keepalive packets report that too? |
| 22:48.29 | ManxPower-work | bcrisp: try the /n option to chan_local |
| 22:48.44 | bcrisp | ok, ill give it a whirl |
| 22:49.13 | [TK]D-Fender | adam0563: I asked you for your DAHDI CONFIGS over an hour ago... |
| 22:50.08 | bcrisp | ManxPower-work: ya, these are using the /n option |
| 22:50.55 | ManxPower-work | bcrisp: then try removing the /n |
| 22:52.36 | *** join/#asterisk fnordus (n=dnall@70.70.0.215) [NETSPLIT VICTIM] |
| 22:52.42 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 22:52.42 | *** mode/#asterisk [+o angler] by ChanServ |
| 22:52.45 | *** join/#asterisk creativx (n=creadure@197.82-134-19.bkkb.no) [NETSPLIT VICTIM] |
| 22:52.49 | *** join/#asterisk mbranca (n=matteo@host139-217-static.224-95-b.business.telecomitalia.it) [NETSPLIT VICTIM] |
| 22:52.50 | *** join/#asterisk b14ck (n=comradeb@cpe-24-24-136-239.socal.res.rr.com) [NETSPLIT VICTIM] |
| 22:52.53 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 22:53.07 | *** join/#asterisk Chinorro (n=Chino@19.226.117.91.dynamic.mundo-r.com) [NETSPLIT VICTIM] |
| 22:53.13 | adam0563 | D-Fender, I was going to get them but someone else started messaging me but then suddenly left |
| 22:53.16 | adam0563 | getting the configs now |
| 22:53.32 | *** join/#asterisk Caplain (i=shayne@caplain.loves.boys.fbi.gov.silverelitez.org) |
| 22:53.49 | bcrisp | weird.. still going straight to voicemail |
| 22:54.35 | bcrisp | Although i dont see the unable to write frame warning |
| 22:54.50 | adam0563 | dahdi-channels.conf -> http://pastebin.ca/1767236 |
| 22:55.44 | *** join/#asterisk tacvbo (n=tacvbo@189.146.183.84) |
| 22:56.01 | adam0563 | system.conf -> http://pastebin.ca/1767237 |
| 23:02.28 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) [NETSPLIT VICTIM] |
| 23:03.28 | adam0563 | http://pastebin.ca/1767246 -> chan_dahdi.conf |
| 23:04.01 | adam0563 | D-Fender, is that all you need? |
| 23:06.19 | *** part/#asterisk btsteve (n=tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
| 23:08.24 | [TK]D-Fender | adam0563: -- Executing [98017266592@default:1] Dial("SIP/100-0000000b", "dahdi/g1/8017266592") in new stack |
| 23:08.37 | [TK]D-Fender | adam0563: all of your DAHDI channels are GROUP=0 |
| 23:08.50 | [TK]D-Fender | adam0563: You have no GROUP=1 channels to dial out of |
| 23:08.55 | [TK]D-Fender | adam0563: that is the problem |
| 23:10.12 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 23:10.22 | adam0563 | oh |
| 23:10.27 | bmoraca_work | just got a RFP to add wifi internet to 20 busses, lol |
| 23:10.59 | bcrisp | interesting... when i removed the t,k options from Dial() it stopped skipping straight to voicemail |
| 23:11.15 | ManxPower-work | bmoraca_work: EVDO/EDGE + Cradlepoint router? |
| 23:11.19 | bcrisp | <PROTECTED> |
| 23:11.37 | bmoraca_work | ManxPower-work: looking at a Cisco 1841 solution...don't trust Cradlepoint much |
| 23:11.41 | ManxPower-work | bcrisp: it would be tk not t,k btw |
| 23:11.53 | bcrisp | yes it was.. just couldn't remember, been a while |
| 23:12.21 | ManxPower-work | bmoraca_work: do they have something that will work with a USB/Express/PCMCIA card? |
| 23:12.30 | adam0563 | D-Fender, how do I change that? |
| 23:12.38 | bcrisp | normally its not an issue, just have oddities showing up when using the queue app |
| 23:12.41 | bmoraca_work | ManxPower-work: they have cellular WICs |
| 23:12.42 | [TK]D-Fender | adam0563: Try a text editor |
| 23:12.46 | ManxPower-work | If so, that would obviously be the right way to go. |
| 23:12.54 | ManxPower-work | bmoraca_work: nifty! |
| 23:12.56 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:13.09 | bmoraca_work | ManxPower-work: Adtran has them, too, but they don't have a router with integrated wifi as far as i can see, and i want a one-box solution |
| 23:13.21 | [TK]D-Fender | adam0563: Or you could try the butterfly approach |
| 23:13.41 | adam0563 | I'll go with that way |
| 23:21.24 | *** part/#asterisk Zaise (n=for@host191-47-static.29-79-b.business.telecomitalia.it) |
| 23:22.12 | *** join/#asterisk tacubo (n=tacvbo@189.146.191.73) |
| 23:25.42 | adam0563 | D-Fender, can I make it call out on group 0 as well as have incoming calls on group 0? I edited the config so that everything is in group 0 and now when it calls out it just sits there and looks like it is going to call out but then does nothing. It shows calling for about 15 seconds and then hangs up. http://pastebin.ca/1767279 <- that is debug |
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| 23:39.33 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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