06:18.21 | *** join/#asterisk infobot (~infobot@rikers.org) |
06:18.21 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.5.0 (2011/07/11), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.12 (2011/07/06) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
06:18.32 | WIMPy | I guess that would make sense. |
06:18.35 | ChannelZ | (from what I understand that setting is really like a 'priority') |
06:18.44 | cusco | right now they are all sequenced anywayz |
06:18.49 | WIMPy | But I have no idea about the dahdi internals. |
06:18.57 | ChannelZ | I wonder if the second card is trying to use sync from the first. Me either. |
06:19.00 | cusco | yes I have been reading it seemslike a priority but .. for the whole system or the rest of the card? |
06:19.17 | cusco | I tried changing it many times |
06:19.18 | cusco | lol |
06:19.24 | WIMPy | ChannelZ: It can't unless there's a timing cable. |
06:19.37 | WIMPy | But it does sond like something. |
06:19.40 | ChannelZ | Hmm. |
06:20.01 | ChannelZ | cusco: have you set up the first 4 spans as 1-4 and the second 4 as 1-4 as well? |
06:20.07 | cusco | ok symptom maintains |
06:20.07 | WIMPy | It IS a priority. |
06:20.15 | cusco | ChannelZ: I did that just now |
06:20.21 | cusco | WIMPy: for the system or the card? |
06:20.39 | ChannelZ | WIMPy: Yeah.. does it matter if the priority starts on "5" for instance? |
06:20.46 | ChannelZ | I guess 0 really only has 'special' meaning? |
06:22.30 | cusco | http://paste.debian.net/124921/ |
06:22.43 | WIMPy | cusco: pardon? |
06:22.50 | cusco | that 'ver-primarios' output is basically a dahdi_scan|grep alarm |
06:23.18 | WIMPy | ChannelZ: I think it's just trying to use the i/f with the lowest configured priority (>0) if possible. |
06:23.35 | WIMPy | But there have definitely been changes iregarding timing source selection in dahdi. |
06:23.37 | ChannelZ | Hmm. |
06:23.54 | cusco | ok let me download older dahdi |
06:23.57 | WIMPy | 0 is never used, yes. |
06:24.02 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
06:24.03 | schmidts | good morning |
06:24.13 | WIMPy | Moin schmidts |
06:24.23 | ChannelZ | And we know the 4 lines running into the card that isn't working work in the card that is. |
06:24.37 | cusco | dahdi-linux-complete-2.3.0.1+2.3.0/ - this is the one I previously had |
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06:25.45 | WIMPy | Or actually 0 is not only not used, but it provides timing. I.e. for NT interfaces. |
06:28.53 | cusco | I think that it being 'internally clocked' is the problem |
06:28.59 | cusco | I'm not sure |
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06:29.54 | ChannelZ | yes but the question is why |
06:30.32 | ChannelZ | you don't have your D channels screwed up do you? |
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06:31.40 | cusco | 34.353023hu? |
06:31.42 | cusco | oops |
06:31.44 | cusco | hu? |
06:31.51 | cusco | Data channels screwed up? |
06:32.03 | ChannelZ | yeah on the wrong channel or something |
06:32.05 | cusco | How would that matter if a span is not connected for instance |
06:32.08 | cusco | lol |
06:32.11 | cusco | no |
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06:32.16 | cusco | genconf gets it right |
06:32.20 | ChannelZ | I'm guessing no since all this worked |
06:32.42 | ChannelZ | well if a span is not connected you're not going to get sync either |
06:32.58 | ChannelZ | Are you saying you're getting all these red alarms with nothing plugged in? |
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06:33.30 | WIMPy | Where does dahdi tell you about 'internally clocked'? |
06:33.44 | cusco | ChannelZ: lol no, one of them has a cable plugged in |
06:33.58 | cusco | WIMPy: dahdi_tool |
06:34.12 | WIMPy | I don't have dahdi_tool :-( |
06:34.12 | cusco | dahdi_scan shows syncsrc=0 |
06:34.31 | WIMPy | Ah |
06:34.42 | WIMPy | That's what I see as well. |
06:34.49 | cusco | where? |
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06:35.30 | cusco | working pri shows synsrc=1 |
06:35.37 | ChannelZ | I don't know why, but that reminds me I need to get my MOH off a backup |
06:35.43 | WIMPy | Looks like dmesg is the only reliable information. |
06:36.12 | WIMPy | I get 0 on a working PRI, as well as an unconnected port. |
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06:37.11 | cusco | dmesg shows messages regarding changing timing sources |
06:37.25 | cusco | because when dahdi starts 1st card spans are yellow, and 2nd card are OK |
06:37.28 | cusco | but only for 3 secs |
06:37.39 | *** part/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
06:37.57 | WIMPy | Which indeed looks like it sees them as one. |
06:38.22 | cusco | http://paste.debian.net/124923/ |
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06:39.05 | ChannelZ | hmm |
06:39.18 | cusco | [ 344.256561] wct4xxp 0000:04:01.0: All spans in alarm : No validspan to source RCLK from |
06:39.24 | cusco | this is my problem |
06:39.29 | WIMPy | There are no messages about sync source on the 2nd card. |
06:39.42 | cusco | WIMPy: the one I just pasted |
06:39.43 | ChannelZ | just out of curiosity, how long have you had this setup with the 2 cards? |
06:39.52 | cusco | ChannelZ: for over a year |
06:40.00 | WIMPy | cusco: Yes, there |
06:40.14 | ChannelZ | and the second 4 always from the same (different) telco? |
06:40.46 | cusco | ChannelZ: yes, and some times more then 1 telco in it |
06:40.58 | cusco | and then I realized I had timing problems and could only keep 1 telco per card |
06:41.40 | cusco | what can I do? |
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06:41.48 | cusco | could it be the card? |
06:41.50 | WIMPy | Which tells us that it would be neccessary to configure timing pools. |
06:41.58 | cusco | several interrupt messages |
06:42.10 | WIMPy | Most probablyt no. |
06:42.11 | cusco | WIMPy: how? |
06:42.26 | WIMPy | cusco: Someone needs to invent something. |
06:42.49 | ChannelZ | I'm wondering if the times it was working was only by accident |
06:42.51 | cusco | but it had been working with no problems |
06:43.02 | cusco | well I restarted it several times |
06:43.05 | cusco | upgraded asterisk |
06:43.09 | cusco | upgraded dahdi |
06:43.10 | cusco | over time |
06:43.11 | ChannelZ | but I don't know enough about the hardcore internals of DAHDI to say |
06:43.15 | cusco | libpri and spandsp |
06:43.24 | cusco | only this time |
06:43.33 | WIMPy | But look at your dmesg. The SPAN x: xxxx sync source only appear for the first 4 spans. That dooes not look right. |
06:43.36 | cusco | I finally moved debian to squeeze |
06:43.46 | cusco | :( |
06:44.22 | cusco | if I coment out span1..4 conf, it will work |
06:44.24 | cusco | I think |
06:44.26 | cusco | let me try |
06:44.29 | cusco | running short on time |
06:44.42 | ChannelZ | Seems like DAHDI as a whole only has one clock |
06:45.14 | WIMPy | That would mean you can't use multiple cards without timing cable. |
06:45.31 | ChannelZ | but he actually needs the timing to be independent since it's from different sources. |
06:45.48 | WIMPy | That would be dahdi--. |
06:45.53 | WIMPy | yes |
06:45.56 | ChannelZ | Unless he can get N-1 of his telcos to sync to him instead |
06:46.09 | WIMPy | No go. |
06:46.12 | cusco | gah now I can't be sure |
06:46.16 | cusco | only 1 cable there |
06:46.20 | cusco | and I can't detect it |
06:46.25 | cusco | its still red |
06:46.39 | cusco | but I could swear that it happened yesterday |
06:46.40 | cusco | or the day before |
06:46.50 | WIMPy | What? |
06:47.14 | cusco | not configuring spans 1..4 |
06:47.26 | cusco | and remaining (from 2nd card) work |
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06:47.41 | WIMPy | Ok, that might make sense. |
06:47.58 | cusco | but I can't be sure now |
06:48.04 | cusco | Will leave it again for tomorrow |
06:48.09 | cusco | service is going to open in notime |
06:48.54 | ChannelZ | I take it it's not saturated |
06:48.57 | WIMPy | Maybe the line was shut down because it's been in alarm? |
06:49.06 | ChannelZ | WIMPy: I have to wonder |
06:49.33 | cusco | the one with the cabel, perhaps |
06:49.38 | ChannelZ | It really seems like the other end is dead or misconfigured or otherwise not cooperating. The card seems unfried |
06:49.39 | cusco | I still have 2 cables disconected |
06:49.50 | cusco | that should be on 1st span |
06:49.50 | WIMPy | Here they tend to disable lines after a certain amount of alarms/time. |
06:49.52 | cusco | they are on |
06:49.56 | cusco | telco is calling me everyday |
06:49.57 | cusco | lol |
06:50.09 | cusco | I have to connect those 2 |
06:51.09 | WIMPy | wonders if the Digium cards would support multiple timing sources at all. |
06:51.19 | WIMPy | The hardware that is. |
06:52.19 | cusco | http://paste.debian.net/124924/ |
06:52.23 | cusco | again |
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06:52.31 | Polysics | hello |
06:52.56 | cusco | hi |
06:53.02 | ChannelZ | Is "IRQ 16/wct4xxp: IRQF_DISABLED is not guaranteed on shared IRQs" normal? |
06:53.10 | Polysics | how would one go about implementing a "credits" system so that a caller has a fixed amount of them, they get removed minute by minute, and finally the call gets dropped if they run out? |
06:53.29 | Polysics | possibly with a "your credit is running out" message, but that is easy given the rest works |
06:53.40 | Polysics | on asterisk 1.8, i have Adhearsion available |
06:53.46 | WIMPy | cusco: Again only 4 timing sources. |
06:53.52 | ChannelZ | Dial has time limits you can set. If a credit is a minute, just feed it the number of credits they have |
06:54.29 | ChannelZ | You ust need to write the back-end to maintain the credit balance, subtracting minutes used after a call, letting them buy more, whatever |
06:54.52 | Polysics | ChannelZ, i know how to d oall that, but the problem is i need to drop calls when credits end |
06:54.57 | Polysics | not let them finish the call |
06:55.11 | WIMPy | And yes, the shared IRQ is definitely not ideal. |
06:55.24 | cusco | Polysics: every time you place a new Dial() you specify the seconts time out flag |
06:55.25 | ChannelZ | I told you, you do a lookup on their 'account' and feed Dial the number of minutes they have. |
06:55.32 | cusco | seconds = credit |
06:56.02 | ChannelZ | whether it's 5 seconds or 5000, it's whatever their balance happens to be at the time they place the call. |
06:56.10 | Polysics | that could work. if the credit does not equal minutes, i can just work out the equivalent |
06:56.20 | Polysics | about the "your credit is running out" item? |
06:56.29 | WIMPy | The interesting bit starts if you let them place more than one call at a time. |
06:56.35 | ChannelZ | Polysics: core show application dial |
06:56.37 | ChannelZ | all is revealed |
06:56.40 | cusco | dial too has a flag for when its timing out, I'm sure I read it |
06:56.53 | cusco | like 15 secs before hangup it plays a warning |
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06:57.13 | Polysics | WIMPy, no, they cannot place more than one call at a time |
06:57.32 | WIMPy | That's fairly easy then. |
06:57.44 | ChannelZ | MusicOnHold needs a flag so people can hit * or # or something to skip to the next song if what they're listening to is crap |
06:57.58 | Polysics | since this thing has a user interface, that shows the credits going down, i guess i could do THAT by "pretend" decreasing the credits, then finalizing them when the call is over |
06:58.11 | ChannelZ | Aren't you listening? |
06:58.18 | ChannelZ | LOOK AT THE L FLAG OF DIAL |
06:58.19 | WIMPy | ChannelZ: Isn't that set by the mode? |
06:58.25 | ChannelZ | It'll annoy them if you want it to |
06:58.34 | ChannelZ | hmm |
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06:59.39 | ChannelZ | WIMPy: not sure I know what you mean. There's no flags for that sort of thing in the app or the config that I am aware of |
06:59.57 | ChannelZ | It would of course make no sense if it was a live stream or whatever, but in files mode.. |
07:00.05 | WIMPy | ChannelZ: I think I've seen something. |
07:00.34 | ChannelZ | I need to change mine, it's all illegal anyway |
07:00.46 | ChannelZ | Nobody gets put on hold that often though |
07:03.11 | ChannelZ | need to spend some quality time on SoundCloud trolling |
07:04.19 | WIMPy | I have no idea, what I was thinking of. But I have no idea where that option might exist, if noch for musiconhold. |
07:04.26 | WIMPy | if not |
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07:05.14 | WIMPy | Ah. Got it. |
07:05.26 | WIMPy | Set digit= in musiconhold.conf. |
07:05.57 | WIMPy | It's for classes, but maybe that's useful. |
07:06.00 | ChannelZ | hmmm I'd never seen that. |
07:06.20 | ChannelZ | That'd work more for like MOH "stations" |
07:06.41 | ChannelZ | like one class of classical, one of pop, that sort of thing. |
07:06.45 | WIMPy | Yes. |
07:06.48 | ChannelZ | Not just "this song sucks, play the next one" :) |
07:07.12 | WIMPy | But maybe it works if you just skip to the same class that uses sort=random? |
07:07.31 | WIMPy | Sounds worth a try. |
07:08.02 | ChannelZ | hmm worth a try |
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07:09.28 | ChannelZ | Interesting. The console tells me 'stopped MOH..' and then 'started MOH...' each time I press, but the song never actually stops or changes, it keeps playing through like I'm doing nothing |
07:10.27 | WIMPy | Bad luck then. |
07:10.50 | WIMPy | But that might be easy to change. |
07:14.03 | ChannelZ | it's a little anecdotal, for another day. Was just testing my MOH after restoring the files (HD died a few weeks ago) and I forgot what some of it even was. |
07:14.26 | ChannelZ | anyway thanks for the kick, I wasn't even aware of the 'digit' option |
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07:17.01 | ChannelZ | Hmm. I didn't know "Skullfuck" was a music classification. |
07:18.52 | ChannelZ | Alright, I should be in bed. Have fun y'all |
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07:26.06 | Polysics | hmmm |
07:26.21 | Polysics | your dial-with-credits idea is great, but i have a problem |
07:26.28 | Polysics | credits are not per user, but per group |
07:26.42 | Polysics | one or more users could be calling at the same time, all drawing credits from the same pool |
07:26.56 | Polysics | so i can't just ask for a timeout, since the timeout is not known |
07:28.46 | WIMPy | The interesting bit starts if you let them place more than one call at a time. |
07:29.02 | WIMPy | That means doing some fancy stuff via AMI. |
07:30.34 | irroot | Wimpy fancy stuff with AMI leads to the nuthouse :P |
07:31.12 | WIMPy | Don't be a wimp :-) |
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07:41.35 | irroot | hehe im nuts |
07:41.39 | irroot | guess why |
07:41.54 | Polysics | WIMPy, what could be a course of action? i will probably use Adhearsion for that part |
07:42.03 | Polysics | the whole app logic is in AHN already |
07:42.34 | Polysics | before i said one call at a time but was only thinking of the single user - the group as a whole can and will do many calls |
07:45.41 | Polysics | i basically would need a kind of timer |
07:45.56 | Polysics | and i hoped to avoid having to build a separate daemon + RPC calls |
07:54.23 | Polysics | any ideas, please? asterisk has nothing like a "call timer" thing? or a periodic event? |
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08:08.46 | Dovid | anyone from France here ? |
08:11.17 | tuxx- | trottoire |
08:11.46 | tuxx- | ehmmm |
08:11.47 | tuxx- | garage? |
08:11.52 | tuxx- | maybe emmmm |
08:11.53 | tuxx- | baquette |
08:11.59 | tuxx- | thats about all i know xD |
08:12.08 | Dovid | lol. i need some one in France |
08:12.20 | tuxx- | hey, just trying to help ;-) |
08:14.11 | Polysics | there is no way to have Asterisk emit a periodic event during a call? |
08:15.23 | Polysics | is there a list of all the AMI events somewhere? |
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08:16.53 | tuxx- | Polysics: http://www.voip-info.org/wiki/view/Asterisk+manager+API |
08:17.02 | tuxx- | check the 'Manager Actions' |
08:17.08 | Polysics | not actions, events |
08:17.12 | Polysics | but thanks |
08:17.22 | tuxx- | ah |
08:17.36 | tuxx- | well, time for a sammich |
08:17.36 | Polysics | i need to somehow get a periodic event out to remove credits fro ma shared pool |
08:17.37 | Polysics | sort of a calling card app |
08:18.12 | irroot | Polysics what we doo is use L option in dial within a AGI script based on database quereies |
08:18.59 | Polysics | irroot, problem is, i cannot know the max duration in advance |
08:19.06 | irroot | i take the avail pool of credit divide by sim use and alocate that ammount as in use then "refund" the ammount left after the call |
08:19.07 | Polysics | because the credit pool is shared between a group |
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08:20.39 | irroot | Polysics take the alloted ammount divide it by rate to work out the limit |
08:22.04 | irroot | this is realtime the downside is you allocate a block of credit for each call that realisticly is not always used up |
08:22.17 | Polysics | so if i hve 5 users in the group, i give total/5 to each maximum? |
08:24.10 | irroot | yip |
08:24.21 | irroot | or say they allowed 5 calls each |
08:24.27 | irroot | then total/25 |
08:25.22 | irroot | i use a "inuse" table where i insert a call and its length into a table and use that to determine num inuse |
08:25.40 | irroot | obviously with the credit allocated to them |
08:26.15 | irroot | select sum(...),sum(credit) from inuse where XXXX = XXXX group by ....; |
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08:38.26 | Polysics | and if a user needs a call to go longer than the allotted time, while having credit for it, and thus he in theory CAN have a longer call? |
08:38.47 | Polysics | say i have 100 credits and 5 users, but only one is calling - will he be dropped at 20 anywya? |
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09:21.29 | teloniusz | hi guys. I'm playing with extensions.conf manually. I'd like to have an extension which just waits indefinitely while the user gets RING signal. How to do it with dialplan functions? |
09:25.00 | gg0 | hi, (another one) how to set a different language just for a specific trunk? |
09:25.02 | teloniusz | oh, I see. Ringing() then Wait() will do the trick |
09:27.10 | dym | teloniusz: yupp |
09:27.41 | dym | gg0: well - in the trunk config |
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09:29.50 | gg0 | I've tried to add language=xx everywhere. I managed to change it just globally in sip.conf |
09:30.06 | Lantizia | is there anyway I can view what password a device is attempting to use? (but failing) |
09:42.42 | teloniusz | gg0: Set(LANGUAGE()=xx) in the dialplan for specific trunk |
09:43.30 | teloniusz | gg0: or better Set(CHANNEL(language)=xx) |
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10:24.57 | volker- | is someone here experienced with sip-tls? |
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12:35.23 | eduzimrs | anyonw kwnows, this message appears at cli : " == Connect attempt from '127.0.0.1' unable to authenticate" |
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12:40.04 | leifmadsen | means something is attempting to authenticate from localhost that is being rejected |
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13:09.52 | eduzimrs | @leifmadsen ok, but trying to auth in a sip peer or * manager? |
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13:10.59 | prash10x | hi |
13:11.53 | prash10x | pls help, i have to configure personal pbxmate |
13:12.20 | prash10x | it is not registering with sip server |
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14:00.01 | E-bola | Isnt there anyway to list configured call/pickupgroups in the asterisk console? |
14:00.11 | E-bola | Or any other way i can get an overview without going through all my sip.conf files |
14:02.48 | *** join/#asterisk darkdrgn2k (~darkdrgn2@199.243.221.14) |
14:02.51 | darkdrgn2k | Hey all, |
14:03.29 | darkdrgn2k | from time to time im getting calls come in with the remote perosn not bein able to hear the person speaking. Any idea what could cause this |
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14:07.22 | *** join/#asterisk joesuffceren (~chatzilla@ip68-104-167-226.ph.ph.cox.net) |
14:07.25 | joesuffceren | Anyone know anything about Cisco 79X0 devices? I have a whole fleet of 7940s and am considering purchasing some 7940Gs. So far as I can tell, the only difference between the two is that the global (7940G) edition uses icons instead of words for the softkeys. I am wondering if I can just flash the 7940G with the 7940 firmware so that they show the words instead of icons. |
14:08.26 | jaytee | darkdrgn2k, make sure your rtp ports set in rtp.conf match what you have open for rtp on your firewall |
14:08.46 | darkdrgn2k | problem seems to go away when we reboot our router.. |
14:08.57 | darkdrgn2k | and its only at that one branch, that would point to a network issue |
14:09.02 | darkdrgn2k | but i have no idea where to even look |
14:10.16 | _Corey_ | joesuffceren: It has to do with the physical buttons on the phone, not the firmware |
14:10.25 | _Corey_ | firmware is the same... |
14:10.58 | _Corey_ | They come with stickers to place around the buttons if you really need the words... most people find the icons enough |
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14:20.39 | joesuffceren | _Corey_: so the screen above the soft buttons would still display text? I couldn't possibly care less about that. haha. Totally misunderstood. Thanks for the clarification! |
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14:21.56 | _Corey_ | Yeah, it's just the four buttons around the (?) button we're talking about on the lower right.... :) No problem |
14:23.56 | p3nguin | G means Global, as in pictures on the face of the phone instead of words. |
14:24.15 | p3nguin | But then you add a sticker to the face overlaying the keys so you have words too. |
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14:40.28 | Qwell | psst |
14:40.29 | Qwell | http://store.digium.com/productview.php?product_code=810-00038 |
14:41.14 | jacc0 | @joesuffceren: I wouldn't buy any cisco or linksys phone right now |
14:41.26 | dvdevel | i have a question that i hope somebody can answer - asterisk 1.6.2.19, if i have a sip entry (say for exten 9705551234) and then i get a call from another asterisk box claiming to be from that number, the call is rejected. can somebody explain why that is, and how to stop that behavior? |
14:42.19 | jacc0 | I've reported some big security hole to cisco yesterday - there is no fix available |
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14:43.05 | jacc0 | I'm not sure the cisco models you are referring to are also effected |
14:43.17 | Qwell | jacc0: oh? |
14:43.29 | Qwell | Is it already publicly known? |
14:43.39 | jacc0 | Uhm, well, ...... |
14:43.49 | jacc0 | you can find some info about it some where |
14:44.15 | *** join/#asterisk DigitalFlux (~quassel@unaffiliated/digitalflux) |
14:44.17 | jacc0 | but it is not yet publicly reported by cisco |
14:44.18 | DigitalFlux | Hi Guys |
14:44.29 | DigitalFlux | I need some caller to input a phone number |
14:44.35 | jacc0 | for now I will not share any technical details |
14:44.42 | DigitalFlux | and i should catch that in some var and Dial() it in my extensions .. |
14:44.47 | Qwell | jacc0: that's fine - was just curious if it was known |
14:44.59 | DigitalFlux | What would be an example for that so that i can put it in my extensions.conf ? |
14:45.01 | Qwell | Once it is released, I'd be interested in hearing about it. |
14:45.04 | jaytee | I've had Beans and Bytes coffee....it was pretty tasty stuff. |
14:45.10 | Qwell | jaytee: Go buy some :p |
14:46.08 | jaytee | Qwell, I'd rather buy some green Kona beans and roast my own. Haven't roasted in almost a year. |
14:46.22 | jacc0 | what I can say is Smithts (not sure what his name is exacly) that hangs around here normaly can confirm the big security hole |
14:47.35 | jacc0 | cisco/linksys phones can be remotely triggerd to ; set up a call, clear call history, update firmwar (about everything you can do if you have the phone on your desk and know the passordw |
14:47.47 | Qwell | this isn't the telnet thing is it? |
14:48.06 | jacc0 | I will not answer to that |
14:48.12 | Qwell | so, yes :) |
14:48.27 | Qwell | if so, it's quite well-known already |
14:48.43 | joesuffceren | jacc0: thanks for the heads up. I have 75 of them in production, though, so one more for a new user isn't going to make or break me. If someone can get telnet access to these phones, they're already far enough inside my netowrk that I'm screwed anyway. :-) |
14:49.50 | DigitalFlux | Guys, how can i capture the numbers from dtmf in a variable in Asterisk ? |
14:50.00 | jacc0 | in some cases a malisous caller id is enough to do the trick |
14:50.00 | DigitalFlux | Any Asterisk app for that ? |
14:50.27 | jacc0 | read() |
14:50.33 | DigitalFlux | checking .. |
14:50.33 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
14:50.48 | DigitalFlux | cool Thanks jacc0 |
14:51.16 | jacc0 | if the phone still has the XSS hole discribed in : uys, how can i capture the numbers from dtmf in a variable in Aster |
14:51.26 | jacc0 | sorry: http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method=fetchBugDetails&bugId=CSCtr27226 |
14:51.46 | jacc0 | sorry again |
14:52.10 | jacc0 | this document : http://www.owasp.org/images/6/6a/OWASPBeNeLux2010-State-VoipHacking.pdf |
14:53.27 | jacc0 | then the new hole can be triggerd by sending a malicious caller id |
14:54.01 | jacc0 | and yes; http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method=fetchBugDetails&bugId=CSCtr27226 is the security report from cisco about it |
14:54.03 | jacc0 | :p |
14:54.05 | Naikrovek | so you weren't going to talk about it but here's the document from last year describing it |
14:54.09 | jacc0 | but it's not public |
14:54.31 | Naikrovek | it's on the internet; it's public |
14:54.31 | jacc0 | combining the bug from last year with this one makes it worse |
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14:55.35 | jacc0 | XSS can be used to attack linebase protocols |
14:55.40 | jacc0 | *line-based |
14:55.51 | jacc0 | as the new bug is in a line based protocol |
14:56.00 | jacc0 | combing the 2 makes it worse |
14:56.59 | jacc0 | you don't even need derect access to the phones ip |
14:57.12 | jacc0 | all you need 2 do is set a malisous caller id |
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14:57.40 | Naikrovek | combining* direct* malicious* |
14:57.48 | Naikrovek | sorry |
14:57.52 | jacc0 | I'm a non native |
14:57.59 | Naikrovek | fair enough |
14:58.27 | jacc0 | @Naikrovek: http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method=fetchBugDetails&bugId=CSCtr27226 is on the internet. so you would say it's public? |
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14:59.32 | Naikrovek | well let me log in and we'll see |
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15:01.00 | Naikrovek | my account won't let me access it, but that doesn't mean it's not public |
15:01.12 | Qwell | just means Cisco doesn't like you |
15:01.20 | jacc0 | lol |
15:01.26 | Naikrovek | i don't have the right type of login |
15:01.35 | Qwell | does |
15:01.36 | Naikrovek | what it means is that i've not whored myself out to them yet |
15:01.39 | Naikrovek | good for you |
15:01.40 | garymc | Hi Guys, for Asterisk ports, do I open UDP 5060 and UDP 10000-20000 in my router firewall to access my asterisk box remotley |
15:01.44 | Qwell | I actually don't. ;( |
15:01.52 | Qwell | garymc: yes.. |
15:01.59 | Qwell | for SIP |
15:02.10 | garymc | is that all I need to make calls? |
15:02.18 | Qwell | sure |
15:02.20 | jacc0 | yes |
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15:05.45 | jacc0 | last word from cisco about the security hole : "Our Dev team had fixed the enclosed issue and will be in the upcoming release." |
15:06.28 | p3nguin | had fixed... as opposed to fixed. |
15:06.39 | p3nguin | That means it used to be fixed. |
15:08.30 | a1fa | is there a website that tells you where number terminates? |
15:08.39 | a1fa | i lost my bookmark, i had one |
15:08.50 | a1fa | it listed the company that owned the number, down to the hub |
15:09.42 | Naikrovek | the answer to your question is "yes" but I don't know the answer to your next question, which is probably "what is the URL" |
15:10.01 | darkdrgn2k | My guess is seeing how you HAD the website at one point it does exist? |
15:10.19 | a1fa | yes please |
15:10.26 | a1fa | there are sites out there |
15:10.30 | a1fa | kind of like domain whois |
15:10.45 | _Corey_ | You looking for the NANPA database of NPA NXX info? |
15:11.25 | a1fa | aybe |
15:11.28 | a1fa | maybe |
15:11.37 | a1fa | i think voip.ms had that feature |
15:11.42 | a1fa | it will tell you who the number belongs to |
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15:11.53 | darkdrgn2k | i havent seen it on their site.. but what do i know |
15:12.19 | *** part/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
15:12.56 | a1fa | does anyone have a voip.ms account? |
15:13.09 | darkdrgn2k | i do |
15:13.13 | _Corey_ | Well, this is where they probably get it from: http://www.nanpa.com/reports/reports_cocodes_assign.html |
15:13.26 | _Corey_ | We have it loaded in MySQL and have an internal portal to run queries |
15:13.41 | _Corey_ | so I can't point you to an easy site, but the data is there |
15:14.05 | jaytee | http://www.area-codes.com/exchange/exchange.asp?npa=513&nxx=322 <-this is what I use for NPA-NXX lookups |
15:14.19 | darkdrgn2k | :( aww no canadian version |
15:14.24 | a1fa | darkdrgn2k: can you check 877-502-6442 |
15:14.30 | _Corey_ | Yeah, that looks like the same data |
15:14.36 | a1fa | Check Availibality on the porting |
15:14.39 | darkdrgn2k | thast an 800 number.. |
15:14.41 | a1fa | it gives you the owner of the number |
15:14.42 | darkdrgn2k | thye never terminate.. |
15:15.34 | a1fa | i wonder how you trace it back to the carrier |
15:15.41 | darkdrgn2k | http://whocallsme.com/Phone-Number.aspx/8775026442 |
15:15.48 | a1fa | yes, i've seen that |
15:16.03 | a1fa | however, i'd like to see who their carrier is |
15:16.49 | darkdrgn2k | i dont know if you can with 8xx numbers |
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15:17.18 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
15:17.32 | a1fa | you should be able to |
15:17.36 | a1fa | i am about to call verizon |
15:17.52 | darkdrgn2k | haha good luck |
15:18.22 | p3nguin | 877..... is an 800 number? Looks like 877 number to me. |
15:18.33 | darkdrgn2k | soryr |
15:18.33 | darkdrgn2k | 8xx |
15:18.40 | a1fa | well, they are harassing my wife on her cellphone |
15:18.48 | a1fa | next thing is android call firewall |
15:18.57 | a1fa | but i'd like to file a complaint with verizon |
15:19.05 | p3nguin | I have a VoIP.ms account, but I don't know how that's going to help you. |
15:19.09 | a1fa | they should not be trunking that number |
15:19.15 | a1fa | p3nguin: if you go check number portability |
15:19.20 | a1fa | it will tell you it cant port the number |
15:19.30 | darkdrgn2k | might be a pw0wned asterisk box |
15:19.30 | a1fa | and it will give you carrier who owns the number to call them |
15:19.43 | a1fa | darkdrgn2k: sure, but it needs to stop, dont you think? |
15:20.21 | a1fa | it does no good to the community thats for sure |
15:20.27 | _Corey_ | alfa: What do you mean by "they should not be trunking that number" ... I'm curious |
15:20.34 | darkdrgn2k | a1fa: for number portability you need to KNOW your own provider |
15:20.44 | a1fa | _Corey_: they should not be routing it on their network |
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15:20.49 | p3nguin | chortles |
15:20.49 | a1fa | darkdrgn2k: not through voip.ms |
15:20.57 | a1fa | it will give you the carrier |
15:21.00 | _Corey_ | alfa: why is that? |
15:21.13 | darkdrgn2k | then voip.ms will not give you the varrier |
15:21.18 | a1fa | _Corey_: it's classified as "spam" |
15:21.34 | _Corey_ | uh yeah, that doesn't matter |
15:21.54 | a1fa | i know it does not |
15:22.03 | a1fa | but if its a scam, dont you think it should be disconnected |
15:22.18 | a1fa | FCC ruled on this once already |
15:22.25 | darkdrgn2k | what are you the us government now? with DID siezeurs :-P hahaa |
15:22.34 | a1fa | ha ha ha |
15:22.48 | a1fa | why is that foreign nationals, outside of US can terminate US DIDs? |
15:23.07 | _Corey_ | I can present that number if I wanted to, as could anyone... |
15:23.07 | a1fa | businesses too |
15:23.36 | darkdrgn2k | yeh cids arent written in stone |
15:23.56 | a1fa | its just bad design, altoughether |
15:24.03 | a1fa | altogether* |
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15:26.02 | a1fa | p3nguin: where you able to check for me, via voip.ms? |
15:26.15 | a1fa | and has anyone seen TK Defender lately? |
15:26.53 | darkdrgn2k | a1fa: where on voip.ms. i dont see it anywhere |
15:27.03 | a1fa | you need to go to number portability |
15:27.07 | a1fa | login to your portal |
15:27.35 | a1fa | http://wiki.voip.ms/article/Porting_a_Number |
15:28.06 | darkdrgn2k | yes |
15:28.07 | darkdrgn2k | i did that |
15:28.11 | darkdrgn2k | but it ask YOUR who the carrier is.. |
15:28.24 | a1fa | try something random |
15:28.26 | p3nguin | I have not checked it, and TK is still around but not on this channel. |
15:28.38 | a1fa | p3nguin: how come? |
15:28.42 | darkdrgn2k | "Service Provider Information" means YOU provide the service providfer info |
15:28.52 | a1fa | type in "Broadvoice" |
15:28.54 | a1fa | hehe |
15:29.06 | p3nguin | When I try to login, it tells me I'm trying from a forbidden IP address. |
15:29.41 | darkdrgn2k | p3nguin, : wouldnt work, voip.ms doesnt tell you the serivce provider anyway.. |
15:29.49 | a1fa | it used to |
15:29.50 | p3nguin | I figured it wouldn't. |
15:29.59 | darkdrgn2k | usualy ports require you to PROOVE you own the # |
15:30.03 | a1fa | i remember trying to port customers number, and it would not do it |
15:30.10 | *** join/#asterisk defswork (~andy@mx2.3gcomms.co.uk) |
15:30.17 | a1fa | darkdrgn2k: aka thats why you click check availability |
15:30.52 | p3nguin | outside of US can terminate US DIDs? <--- what does this even mean? |
15:30.59 | p3nguin | DIDs don't terminate. |
15:31.14 | a1fa | sip trunks |
15:31.19 | p3nguin | no such thing |
15:31.27 | a1fa | .. ok sherlock |
15:31.33 | p3nguin | Yes, Watson? |
15:31.40 | a1fa | don't beat around the bush |
15:31.49 | darkdrgn2k | a1fa,: Your absolulty right.. but TOLL FREE numbers DONT HAVE check availabllity! |
15:31.55 | a1fa | what's D-Fender's nick? |
15:32.01 | a1fa | darkdrgn2k: thanks for checking |
15:32.02 | defswork | I was getting repeatable deadlocks in queue so I took all my queues out thinking that would solve it but have had another deadlock today. Is 1.8.5 stable ? Am I the only one having these kind of problems ? |
15:32.07 | p3nguin | Termination is for OUTbound calls. DIDs are INbound. |
15:32.15 | jaytee | a1fa, go to the #freepbx channel if you want to chat with TK |
15:32.16 | p3nguin | [TK]D-Fender |
15:32.23 | a1fa | p3nguin: hanks for setting me straight |
15:32.39 | drmessano | origination is inbound |
15:32.42 | drmessano | errr |
15:33.16 | p3nguin | When a call comes into your DID from the PSTN, that's origination (not termination). |
15:33.34 | a1fa | p3nguin: thanks |
15:34.20 | p3nguin | If toll-free numbers don't have the "check availability" thing, does that mean they don't port them? |
15:34.27 | darkdrgn2k | what if it originates from somethign other thebn a PSTN like skype :) |
15:34.37 | darkdrgn2k | p3nguin: no 800 numbers can ALWAYS be ported because of their nature |
15:34.59 | darkdrgn2k | p3nguin: in the PSTN world an 800 number MUST be attached to a normal DID |
15:35.00 | p3nguin | They just can't check it because there is no "declined" choice? |
15:35.17 | p3nguin | What do you mean attached to a normal DID? |
15:35.34 | darkdrgn2k | you cannot have a PAIR with only an 800 numbe ron it |
15:35.40 | p3nguin | I have a toll-free DID, and that's all that I have. It seems normal to me. |
15:35.49 | darkdrgn2k | is it a voip line? |
15:35.53 | p3nguin | yes |
15:35.57 | darkdrgn2k | as i said |
15:36.07 | darkdrgn2k | voip broke that requirement |
15:36.16 | darkdrgn2k | in the OLD PSTN world, you could not have a dial tone with only an 800 numbe ron it |
15:36.17 | p3nguin | I didn't see you say that. |
15:36.40 | darkdrgn2k | sorry i guess i thought it :( |
15:37.17 | darkdrgn2k | but yeh you cant get a pair come into your office with only an 800 number on it.. the dmark is always a local line... |
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15:37.20 | p3nguin | So if I go back to copper, I will have to have a "regular" number before I can port my toll-free over? |
15:37.27 | darkdrgn2k | thast why 800 numbers are so portable |
15:37.27 | darkdrgn2k | yep |
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15:37.48 | p3nguin | Just like if I have only DSL... there is still a phone number on the circuit even if I don't have a phone. |
15:38.07 | elb | and it confuses the @!$!@$! out of support drones |
15:38.16 | a1fa | +1 elb |
15:38.17 | a1fa | +1 |
15:38.18 | elb | "I show the number on your account as 111-222-3333" |
15:38.32 | a1fa | numbers are cheap |
15:38.37 | darkdrgn2k | worst part is you CID wont show the 800 numers either :) |
15:38.38 | a1fa | lets toss them around like salad |
15:38.41 | elb | "no, the number on my account is <some real number>, you can't call me at that number, it has no POTS line" |
15:38.53 | elb | "but I have to call you at the contact number" |
15:38.55 | elb | "@#$@#$@#$" |
15:39.21 | p3nguin | How do you ever get them to move past that and get to the actual problem? |
15:39.28 | a1fa | you dont |
15:39.33 | elb | p3nguin: call back and get a different rep |
15:39.49 | elb | the best part is when they ask you for the number, and you don't know it, because they've never given it to you |
15:39.50 | defswork | is anyone else aware of deadlocking problems in 1.8.5 ? |
15:40.12 | elb | I gave all those problems up, though ... now I have VDSL (AT&T U-Verse) without television service, which is always the sticking point |
15:40.14 | p3nguin | Is that when you call back and get someone else to give you the number? |
15:40.22 | elb | they swear they don't sell U-Verse without TV service when I call |
15:40.24 | elb | and won't help me |
15:40.30 | elb | telcos are AWESOME |
15:40.34 | darkdrgn2k | the best part is whne they ask you for the phone number, then the accout number, which is the same, and they insist its not |
15:40.44 | darkdrgn2k | im like IM LOOKING AT THE BILL.. IT SAYS ACCOUNT NUMBER RIGHT HTERE!!! |
15:41.29 | darkdrgn2k | i have VDSL:( 25 megabit connection with a 75 gig cap.. WTF! |
15:41.39 | darkdrgn2k | Lets do the math. thast what 4 ours of downloading |
15:41.41 | p3nguin | I don't have an AT&T bill handy, but I thought the phone number is the same as the account number. |
15:41.43 | a1fa | ATT UVERSE? |
15:41.55 | a1fa | lol 75 gig cap? |
15:41.56 | darkdrgn2k | p3nguin: it is with BELL.. |
15:41.58 | darkdrgn2k | yep |
15:42.04 | a1fa | i cant even fart with 75 gigs |
15:42.12 | darkdrgn2k | they they are like We can upgrade you to the 50 megabit ... |
15:42.14 | a1fa | i'd be running out daily ;) |
15:42.16 | darkdrgn2k | im like whats the cap on that |
15:42.17 | darkdrgn2k | 100... |
15:42.21 | a1fa | ROLF |
15:42.34 | darkdrgn2k | im like " send that person back to school, he needs to learn fractions" |
15:42.48 | darkdrgn2k | the person on the phone said i can buy INSTAUCACE at 5 bucks a pop x 3 to boos me up to 350.. |
15:42.53 | darkdrgn2k | thatst 24 hours of downloading i think |
15:42.53 | p3nguin | I'm sure that's not the only area which needs improvement. |
15:43.50 | a1fa | nah |
15:43.59 | a1fa | you'll be far fetched to use 75 gigs |
15:44.06 | darkdrgn2k | Yeh |
15:44.09 | a1fa | unless you are on torrents all day |
15:44.12 | a1fa | 24/7 |
15:44.15 | darkdrgn2k | umm |
15:44.16 | darkdrgn2k | dude |
15:44.18 | a1fa | and even then.. |
15:44.23 | darkdrgn2k | i upload 4 gigs ISOs to the datacenter.. |
15:44.29 | darkdrgn2k | and stuff |
15:44.35 | darkdrgn2k | and its 75 gigs UP + DOWN not UP OR DOWN |
15:44.37 | a1fa | have your work pay for the fees associated with overages |
15:44.51 | darkdrgn2k | so download the ISO from eopen... 4 gigs.. upload it to the datacenter .. 4 gigs.. |
15:45.07 | a1fa | why do you upload isos? |
15:45.17 | darkdrgn2k | ESXi.. |
15:45.22 | a1fa | but why |
15:45.31 | darkdrgn2k | cuase i have no GUIs at the datacenter |
15:45.36 | a1fa | wget? |
15:45.37 | darkdrgn2k | and eopen doesnt work with lynx of wget |
15:45.49 | a1fa | so put a jump box over there |
15:45.56 | darkdrgn2k | no room:( |
15:46.02 | darkdrgn2k | anywa moot point |
15:46.03 | a1fa | make a virtual jump box :) yo |
15:46.12 | darkdrgn2k | download 2 bluerays.... and BOOM... there goes 60 gigs |
15:47.02 | darkdrgn2k | in december before the new rules cam into affect (last year it was 75 gigs. buck a gig after that to a max of 30 bucks) i tried to see what how much i could pull in a month.. just a proof of concept. |
15:47.12 | darkdrgn2k | i ended up pulling like 1.7 tb.... and ran out of hd space |
15:47.16 | darkdrgn2k | then i did the match |
15:47.49 | darkdrgn2k | 1700 gigs = 75 gigs free + 250 gigs "insured" leaves 1375 gigs x 1 buck a gig = $1,375 |
15:48.00 | darkdrgn2k | i cant WAIT for a virus to infect one of theses poor suckers and rape their bandwith :) |
15:48.10 | a1fa | ;) |
15:48.13 | p3nguin | I have a Windows XP ESXi machine for that sort of thing. It's kind of a bother, but it works. Since I don't use Windows on the desktop, and the console uses Windows... rdesktop to the XP vm and control all the other vms. |
15:48.24 | a1fa | sounds like I'll be able to make a living :) |
15:48.43 | darkdrgn2k | lol |
15:49.11 | a1fa | at $79.99 for diagnostics, thats bread and butter.. free money |
15:49.19 | a1fa | $175 to fix simple things |
15:49.42 | darkdrgn2k | lol |
15:49.44 | darkdrgn2k | still |
15:49.52 | darkdrgn2k | why have vdsl when you have a 75 gig cap |
15:50.00 | darkdrgn2k | im terrified to do stuff at home now |
15:50.56 | a1fa | cancel |
15:50.58 | a1fa | go buy cable |
15:51.09 | darkdrgn2k | i'd miss the 8megabit upload! |
15:51.21 | a1fa | you need to find alternative means dude |
15:51.25 | elb | cable is stupid expensive here |
15:51.31 | elb | of course, my vdsl is also slow |
15:51.44 | a1fa | $49.99 here with 12Mbit/3Mbit d/u |
15:51.48 | elb | but ... I can't get cable for less than about $60/mo |
15:51.56 | coppice | my kids can burn through 75G of youtube in no time at all |
15:52.14 | darkdrgn2k | i love it |
15:52.23 | darkdrgn2k | every one is doing the "TO THE CLOUD" krap.. |
15:52.31 | darkdrgn2k | and isps are sayinf PAY US FOR THE CLOUD |
15:53.01 | elb | the cloud is a lie |
15:53.23 | darkdrgn2k | meh i loke the cloud |
15:53.29 | darkdrgn2k | at least the ons hosting my files at home :) |
15:53.34 | darkdrgn2k | another reason i LOVE vdsl |
15:53.35 | coppice | the cloud is very honest - its a bit wet |
15:55.40 | p3nguin | Miss the 8 Mb upload? Why? I get damn near that much on cable. |
15:56.38 | darkdrgn2k | we dont |
15:56.44 | darkdrgn2k | ours is is like 4 max |
15:57.56 | p3nguin | We also have 100 Mbit download speeds on cable. |
15:58.06 | a1fa | so why did d-fender get shit canned? |
15:58.39 | p3nguin | I don't know about shit-canned, but he got a +q for saying a lot of unnice things. |
15:59.14 | p3nguin | Eventually he just quit coming here reading what others said. |
15:59.56 | darkdrgn2k | yeh... but hes fun to watch some times :-P |
15:59.59 | p3nguin | I don't really think anything he said was uncalled for. But I've been told I'm not nice at times as well. |
16:00.00 | a1fa | suprising, he was always nice? |
16:00.12 | a1fa | and always helpful |
16:00.20 | jaytee | p3nguin, you're always mean and nasty :-) |
16:00.21 | beek | His banishment has been a major loss to this channel. |
16:00.31 | coppice | he made the fatal mistake of being helpful and honest. only do a maximum of one at a time |
16:00.48 | darkdrgn2k | its funny it hoguth he'd leave freepbx first... he hats the crazuy dialplans |
16:00.48 | a1fa | p3nguin: you just need to be more precise, and stop beating around the bush. if somebody is not correct, just say it out right, and correct them |
16:00.48 | jaytee | p3nguin, while being extremely helpful at the same time |
16:00.52 | p3nguin | Regardless of his style, he did help a lot of people here. |
16:00.53 | a1fa | no need to point fingers and laugh |
16:01.12 | a1fa | beek: i agree.. that's what suprised me |
16:01.17 | darkdrgn2k | umm guys |
16:01.20 | darkdrgn2k | you know this is IRC right? |
16:01.26 | a1fa | and? |
16:01.26 | darkdrgn2k | pointing an lafing is like a GIVEN! |
16:01.39 | a1fa | not on freenode, it has never been like that |
16:01.47 | a1fa | a lot of things have changed in last few years |
16:01.49 | darkdrgn2k | LMAO where have you been? |
16:02.05 | a1fa | darkdrgn2k: i've been on here since the network has been started |
16:02.06 | darkdrgn2k | ok maybe not POINT AN LAF but smartassing is the LAW |
16:02.14 | p3nguin | I guess if you all want him back, you could always start a petition and give it to russelb. |
16:02.24 | a1fa | the attitude has changed dramatically in the last few years back |
16:02.36 | darkdrgn2k | the internet has changed.. |
16:02.38 | beek | p3nguin: russellb now works for Red Hat |
16:02.46 | a1fa | freenode used to be a place to get away from assholes on other networks, and actually get help |
16:02.51 | a1fa | looks like trolls migrated |
16:02.54 | p3nguin | He doesn't manage this channel anymore? |
16:03.07 | beek | I wouldn't think so. |
16:03.12 | darkdrgn2k | a1fa: i always get my ass wooped.. but lear sometihngi n the process... |
16:03.13 | p3nguin | I did not know that. |
16:03.14 | Qwell | He's still an op. |
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16:03.43 | p3nguin | He's the one that apparently made the decision about fender. |
16:03.45 | beek | Makes more sense for a Digium employee to be deciding what happens on this channel. |
16:04.00 | beek | p3nguin: yes. I saw the exchange. 'fender sent me the links to the logs. |
16:04.07 | a1fa | darkdrgn2k: rewind that few years back |
16:04.13 | a1fa | you could actually learn something here |
16:04.23 | Kobaz | oh, is that why fender is awol |
16:04.30 | philippel_mac | question, anyone know if there is a counterpart to the IMPORT() function, or ImportVar application, e.g. the ability to set variables in another channel? |
16:04.32 | a1fa | there are few remaining channels left that are actually good |
16:04.41 | a1fa | everything else is a hit or miss |
16:04.46 | philippel_mac | basically something like MASTER_CHANNEL() but where the channel can be specified? |
16:04.58 | p3nguin | a1fa: Society keeps deteriorating, so those of us who are intolerable get noticed more now. |
16:04.59 | a1fa | ever since LILO died, the network has been going down hill |
16:05.01 | philippel_mac | in 1.8, or otherwise a patch or in trunk that could be back ported? |
16:05.18 | a1fa | lilo used to gline those trolls, pretty quick |
16:05.21 | darkdrgn2k | i been trollin since the days of multi collide bots and REAL net splits.... the ppl got more trollier (on avg) but the IQ has just increasesd. if you knwo what ot ask, what to take seriosuly and what NOT to do to feed the trolls its a great place |
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16:05.35 | beek | Well, 'fender may have been the rudest guy on the channel but if you did as he asked and provided the info that he asked he could have your problem solved very quickly. He was extremely helpful to me. |
16:06.04 | darkdrgn2k | agreas with Beek |
16:06.07 | a1fa | he was never rude to me, and always helped |
16:06.14 | a1fa | going back 5 years |
16:06.20 | Kobaz | i always found fender amusing |
16:06.23 | beek | alfa: he just didn't suffer fools well. |
16:06.24 | a1fa | or actually more than 6 years |
16:06.26 | darkdrgn2k | a1fa: he has no patients for NOOBs thast the problem |
16:06.34 | a1fa | no body does |
16:06.40 | p3nguin | some less than others. |
16:06.45 | darkdrgn2k | a1fa: yet every one of us had been one at one point |
16:06.58 | Qwell | I was never a noob. |
16:07.02 | Qwell | That is a documented fact. |
16:07.09 | a1fa | i was born with a 12 inch cock, but that's just me |
16:07.10 | beek | He just followed the same rules on IRC as applied to mailing list: DO YOUR HOMEWORK FIRST. |
16:07.11 | darkdrgn2k | but as a noob i took his critisizm to heart and he helped me crack Nortel's SIP config for freepbx :) |
16:07.11 | Qwell | leifmadsen: wait, was it noob or newb? |
16:07.19 | philippel_mac | Qwell: any idea on the above setting channel var in another channel? |
16:07.35 | Qwell | philippel_mac: I don't. Tilghman would be a good person to ask though. |
16:07.37 | darkdrgn2k | r/noob/nweb/ |
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16:09.50 | leifmadsen | Qwell: nub |
16:09.57 | leifmadsen | Qwell: oh right, newb |
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16:16.39 | Kobaz | who'se going to be the new russell? |
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16:22.03 | serafie | Kobaz: right now three or four existing Digium people are divying up the work, including kpfleming and The_Boy_Wonder |
16:22.53 | The_Boy_Wonder | Kobaz: i'm looking at the timerfd issue now |
16:23.09 | The_Boy_Wonder | where you ever able to reproduce it consistently? |
16:23.57 | The_Boy_Wonder | s/where/were |
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16:40.25 | Kobaz | The_Boy_Wonder: yeah i can reproduce it pretty often with my unit testing |
16:40.40 | Kobaz | The_Boy_Wonder: but i haven't put together a sample dialplan that does it... i have a lot of nuttyness going on |
16:41.53 | The_Boy_Wonder | Kobaz: if there is anyway you could narrow it down to something I can easily do here, that would be awesome for testing this. I have a few ideas on what might be going on. The problem is the turn around time required to tell if experimental code makes a difference or not |
16:42.12 | The_Boy_Wonder | if i can reproduce it here, i can get this done much quicker |
16:42.47 | The_Boy_Wonder | otherwise its a process of posting a patch, and waiting for feedback |
16:43.20 | Kobaz | yeah |
16:43.24 | Kobaz | i know it's brutal |
16:43.40 | Kobaz | I'm currently breaking everthing in my development, so it'll be a little bit |
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16:45.10 | The_Boy_Wonder | Kobaz: alrighty |
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17:02.18 | dvdevel | with asterisk 1.6.2.19, if i have a sip entry (say for exten 9705551234) and then i get a call from another asterisk box claiming to be from that number, the call is rejected, even if it's to another valid number in that box. can somebody explain why that is, and how to stop that behavior? |
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17:08.07 | ChannelZ | See the 'naming devices' section of the sample sip.conf |
17:13.09 | dvdevel | yes, that makes some sense. still, why would it reject the call - it's a valid source of calls if it exists. |
17:13.58 | dvdevel | ah, because the credentials don't jive |
17:14.34 | dvdevel | allow me to think on this. thanks, ChannelZ |
17:15.43 | ChannelZ | yes if it's matching the call to a peer when it shouldn't. Using extensions/numbers as device names is probably not a good idea in your case |
17:16.40 | dvdevel | aye, i think i can work around it "easily enough" |
17:18.34 | citywok | dvdevel: it would be really annoying if random calls could be sent to all your 4 digit extension desk phones and drive all your people crazy |
17:18.48 | citywok | also, how would asterisk know what context to send those calls to without a peer definition telling it? |
17:19.16 | dvdevel | agreed - i just wasn't "putting two and two together" that it wasn't matching the trunk entry but rather the extension |
17:19.28 | ChannelZ | the whole user/peer/friend thing and what happens is still clear as mud |
17:19.31 | dvdevel | i mean i've _only_ been using asterisk for about five years. |
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17:23.24 | azv4 | I know this is OT, but I know there are some old phone system pros around here! Any Panasonic Digital Hybrid phone pros remember if it is possible to connect to all phone's speakerphone in case of an emergency? |
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17:49.49 | nny | is there a way to playback a tone in a meetme to only one side? |
17:49.55 | nny | vs announce, etc |
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17:50.45 | nny | can be pre the other person joining, I can use a hackish command to play something in that room before the join |
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17:58.00 | eduzimrs | anyonw kwnows, this message appears at cli : " == Connect attempt from '127.0.0.1' unable to authenticate" its a sip or manager connection type? |
17:58.01 | nny | nm got it :D |
17:59.18 | ChannelZ | eduzimrs: Looks like Manager |
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18:02.45 | neurosys | leifmadsen: Like breaking from a queue, can a dialplan be made to break from music on hold to leave a VM if the caller is placed on hold and decides they no longer with to hold and leave a VM? |
18:03.56 | leifmadsen | neurosys: application map in features.conf |
18:04.16 | neurosys | leifmadsen: Looking. Thanx :) |
18:04.23 | *** join/#asterisk rjune (~rjune@75-150-213-153-Illinois.hfc.comcastbusiness.net) |
18:05.55 | p3nguin | I use a short queue timeout with a prompt to leave a voicemail if they want... or continue to hold, which drops the call back into the queue. |
18:06.35 | rjune | "called g0/#######" in the log indicates asterisk has picked up a line and dialed the number, correct? |
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18:07.06 | p3nguin | I guess your moh would have to play a message to the call on hold for the person to know he can press a key to leave a message. Most people sitting on hold think they have only two choices: wait longer or hang up. |
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18:22.49 | NephFL | I have a server randomly rebooting with no errors showing in logs, dell t110, have run bios and raid updates...have switch power supply mb and cpu... have digium AEX800 to connect to incoming pots ... |
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18:23.26 | NephFL | running on CentOS 5.5 (freepbx distro)... and short of rebuilding...I'm at a loss |
18:25.46 | rjune | NephFL, can you take it down temporarily? |
18:26.25 | NephFL | its live |
18:26.34 | rjune | I understand, is it a 24/7 shop? |
18:27.26 | NephFL | no, 730 to 5 i think |
18:27.36 | NephFL | what do you have in mind? |
18:29.44 | rjune | RAM test in specific |
18:29.54 | rjune | bad hardware will do what you're seeing |
18:30.14 | rjune | Run memtest86 on it, |
18:30.25 | rjune | Inquisitor seems to be a decent hardware test suite in general |
18:30.33 | rjune | Just be careful not to wipe the drive |
18:34.47 | eduzimrs | ChannelZ should manager try to connect from localhost? |
18:35.13 | eduzimrs | ChannelZ it never happend before |
18:37.32 | neurosys | p3nguin: and how would you define the hold to go to a queue as opposed to the hold app? |
18:38.36 | p3nguin | You don't. That's not what queues are for. |
18:39.25 | neurosys | Ok. I understand the queue break out... |
18:39.52 | neurosys | But the customer wants "If they are place on hold after pickup, the ability to press # and leave a message and hangup". |
18:40.05 | neurosys | I think Ill just tell him not possible :P |
18:40.30 | neurosys | Dang customers with their weird requests :P |
18:44.33 | p3nguin | I don't know if it's possible or not. Dial()'s d option is similar, but it indicates that it is to be used while waiting on the call to be answered. If the call is on hold, it has already been answered. |
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19:08.20 | Gokee2 | Hello everyone, I have a Digium 410p PCI card with two FXS and two FXO ports. After moving it to a new computer the fxs ports have stopped working. However in asterisk the ports are all seen and appear to work, they are just dead when you plug a phone in. Any idea's? |
19:08.38 | *** join/#asterisk Beltechs (~Beltechs@cpe-76-175-74-169.socal.res.rr.com) |
19:09.35 | JonathanRose | Gokee2: pastebin your chan_dahdi.conf and your dahdi/system.conf |
19:09.52 | JonathanRose | That'll help someone to take a look. |
19:10.52 | JonathanRose | I'm pretty sure the card I'm using is similar. |
19:10.57 | Beltechs | hello, Im running * 1.6, sip trunk, G729 codec, I'm having random calls with static. Including the initial ring produced by the pbx. Its most noticeable when dialing 800#'s any ideas would be appreciated. |
19:11.13 | chazzam | Gokee2: is the power cord on the board connected? |
19:12.01 | Gokee2 | chazzam, Hey, good question! I had not though to ask that yet. |
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19:16.31 | chazzam | Beltechs: the SIP phone itself produces the ringing sound generally, not the PBX. if you get static there, then its probably the phone. |
19:16.42 | ChannelZ | eduzimrs: well people or other apps connect to manager, and could have tried to do so to localhost. You'd have to figure out what actually triggered it |
19:17.01 | ChannelZ | Possibly just a guy doing 'netstat' on your box and seeing that port being listened to, and telnetting to it or something. |
19:19.10 | a1fa | Qwell |
19:20.02 | Gokee2 | JonathanRose, Ok here is the chan_dahdi http://pastebin.com/dNasuQzF dahdi-channels.conf http://pastebin.com/ngzQFeDZ and system.conf http://pastebin.com/zrXAKYNC Let me know if you see anything wrong. But I think chazzam may be right with his unplugged comment... Got to contact the site again to find out though |
19:21.12 | rjune | "called g0/#######" in the log indicates asterisk has picked up a line and dialed the number, correct? |
19:22.18 | *** join/#asterisk PoWeRKiLL (~powerkill@bzq-79-183-200-176.red.bezeqint.net) |
19:22.26 | PoWeRKiLL | hi |
19:22.31 | PoWeRKiLL | coppice are you there ? |
19:23.01 | chazzam | Gokee2: that system.conf only configure ports 1-3, while asterisk is configured for 1-4 |
19:23.41 | chazzam | asterisk should fail to load chan_dahdi |
19:24.00 | coppice | PoWeRKiLL: possibly |
19:24.16 | PoWeRKiLL | coppice I got a fax problem |
19:24.21 | PoWeRKiLL | I have a WARNING T.30 Page did not end cleanly |
19:24.44 | PoWeRKiLL | I made a pcap of all the call |
19:25.01 | coppice | T.38? |
19:25.10 | PoWeRKiLL | Yes |
19:25.40 | coppice | could be a broken implementation at the far end. does he page look OK? |
19:25.58 | PoWeRKiLL | The start of the page yes |
19:26.04 | PoWeRKiLL | then it's corrupted |
19:26.38 | coppice | send me the pcap and I'll take a look |
19:27.50 | Gokee2 | chazzam, Interesting, thanks! Fixed http://pastebin.com/u2EKyT5P |
19:28.46 | ChannelZ | rjune: mostly yes |
19:29.20 | ChannelZ | rjune: it doesn't necessarily mean it succeeded AFAIK |
19:30.48 | JonathanRose | It could also just be configured backwards. Here's mine, using Wildcard TDM410P, and it works: http://pastebin.com/tnuhYe6K |
19:30.58 | JonathanRose | Gokee2: poke |
19:31.22 | JonathanRose | I think the auto-generated script might have been backwards for me at first by the by. |
19:32.11 | ChannelZ | It just depends on the order the modules are installed |
19:33.00 | ChannelZ | And remember an 'FXO port' uses FXS signalling and vice versa, so the config might always seems backwards from how you think of it |
19:33.35 | JonathanRose | Yeah, but I mean... if he's using FXS on FXS, it won't work, and everything will look right, but he wouldn't get dial tone. |
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19:34.31 | JonathanRose | I'm no dahdi guru though. |
19:35.15 | Gokee2 | JonathanRose, Its the correct order. The first two ports work, and its worked with that config up until the card was moved. |
19:36.13 | JonathanRose | that seems pretty reasonable then. |
19:36.37 | Gokee2 | I am betting the power was just never plugged in when it was moved |
19:37.02 | Gokee2 | Not sure though, should that kill the dialtone as well as ringing? |
19:37.13 | JonathanRose | Yes |
19:37.26 | Gokee2 | Ok :) |
19:37.54 | JonathanRose | I think the power is basically just used to amplify the outgoing signal. |
19:38.02 | Gokee2 | Ah |
19:38.46 | rjune | ChannelZ, I just needed to know if that meant it tried |
19:38.54 | rjune | looks like channel 1 has no dial tone |
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19:47.15 | asterisk978 | Upgrading from 1.6.2 to 1.8.5 causes issues with my dialplan when using CDR(accountcode) and then dialling a local channel you can not access the accountcode. This worked in 1.6.2 but nolonger in 1.8.5. Is this a bug? |
19:50.37 | jeffspeff | does res_fax.so and res_digium_fax.so conflict? |
19:51.22 | p3nguin | That reminds me I still haven't fixed my fax support. |
19:51.49 | Qwell | jeffspeff: The latter requires the former. |
19:52.13 | asterisk978 | sorry should have said when using SET(CDR(accountcode)=23244) |
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19:52.20 | jeffspeff | hmm... thanks Qwell |
19:52.55 | jeffspeff | Qwell, do you have a second to lend a hand with fax detection not working right? |
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19:59.01 | asterisk978 | can any one help? I raised this as a bug, however it was closed |
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20:01.50 | diegocn | hello ppl... can asterisk be used behind a proxy like TOR? |
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21:04.24 | p3nguin | loader.c:382 load_dynamic_module: Error loading module 'res_fax_digium.so': /usr/lib/asterisk/modules/res_fax_digium.so: undefined symbol: ast_fax_state_to_str |
21:04.29 | p3nguin | What's the fix for this? |
21:07.00 | p3nguin | This is asterisk 1.4.39.2, res_fax-1.4_1.3.0-x86_32, and res_fax_digium-1.4_1.3.0-i686_32. |
21:10.43 | p3nguin | I guess I'll try res_fax_digium generic instead of i686. |
21:13.59 | p3nguin | That didn't help. |
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21:21.05 | p3nguin | It looks like using preload => res_fax.so helps res_fax_digium.so to load correctly. I guess that indicates some sort of race condition during loading of modules. |
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21:22.43 | chazzam | JonathanRose: Gokee2 power is required to the board for FXS ports to work. period. They cant draw the required power from the pci bus |
21:23.29 | chazzam | s/2/2:/ |
21:23.35 | chazzam | I forgot that did that... |
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21:44.41 | radic | is there a way to place a call from the CLI und redirect it to a SIP-phone? |
21:45.15 | p3nguin | redirect, no. Connect it to, yes. |
21:45.24 | p3nguin | CLI, originate |
21:46.19 | p3nguin | Do you want the call to first hit the SIP phone and then ring out to the other phone, or do you want it to ring out to the other phone and if someone answers then give it to the SIP phone? |
21:46.39 | chazzam | I just want cookies |
21:46.41 | radic | p3nguin: the second |
21:47.31 | p3nguin | Do you have extensions configured to call out to that number and also to call the SIP phone? |
21:48.19 | p3nguin | If so, use something like this: originate Local/3149691077@outbound_calls extension 3001@phones |
21:48.54 | p3nguin | If not, use something like this: originate SIP/valid-peer-here/3149691077 extension 3001@phones |
21:49.23 | p3nguin | If the first part of that gets an answer, it will call extension 3001 in the phones context (your SIP phone's extension). |
21:50.10 | p3nguin | valid-peer-here is, of course, your ITSP peer name as configured in sip.conf. |
21:51.20 | radic | hmm |
21:52.15 | p3nguin | I know that "hmm." What part are you having trouble with? |
21:53.03 | chazzam | I think he wants cookies now |
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21:55.23 | radic | p3nguin: I'm not at home an there is only a phone for incoming calls and I want that asterisk calls the number and If a recive the call it let ring for example the SIP-phone withe the extension 374 |
21:56.05 | p3nguin | Can you repeat that again in plain English, please? |
21:57.30 | p3nguin | You asked how to make a call from the CLI and connect it to a phone. I provided you with that information. |
22:02.20 | radic | p3nguin: and 3149691077 is the number that should be called in your example? |
22:03.26 | p3nguin | correct |
22:04.22 | p3nguin | If you already have an extension to call out, I'd use the Local channel to call out. See first example. |
22:04.32 | *** part/#asterisk dvdevel (~devel@wiggum.digitalcoven.com) |
22:06.13 | radic | p3nguin: and If I take up the reciver (in the 2nd example) it calls 3001 in the extension phones? |
22:07.21 | p3nguin | Both examples will call extension 3001 in the phones context if the outside number called answers. |
22:08.12 | p3nguin | Local/3149691077@outbound_calls utilizes a preconfigured extension in a context by the name of outbound_calls. If your context is something else, change it. |
22:08.27 | p3nguin | If you don't wnat to call 3149691077, change that. |
22:08.49 | p3nguin | If your SIP phone's extension isn't 3001 in context phones, change that, too. |
22:09.08 | p3nguin | It's an EXAMPLE. You can surely figure out what bits need changed for your real usage. |
22:09.25 | radic | I'll move to the phone and tray it... |
22:10.31 | p3nguin | Unless you tell me what number you want to call, what context your outbound calls are going through, what SIP peer you call out of, the extension number of your SIP phone, and what context the SIP phone's extension is in, I can't write the exact literal copy/paste command for you. |
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22:27.37 | radic | hmpf |
22:27.55 | radic | all phones here arn't working... |
22:28.01 | radic | +e |
22:28.25 | *** join/#asterisk talntid (~erict@li93-153.members.linode.com) |
22:28.46 | talntid | Asterisk died with code 1. |
22:29.06 | talntid | logs aren't giving any errors about it.... any ideas what usually causes that? dialplan error? |
22:42.32 | radic | talntid: what did you before asterisk died? |
22:43.51 | talntid | this is a customer that called me |
22:44.01 | talntid | but supposedly... http://www.withsupport.co.uk/node/83 |
22:44.24 | WIMPy | talntid: I haven't seen dilpaln errors causing serious truble in operation. but they can cause crahses on reload. |
22:44.39 | talntid | i have returned both of those modified files back to factory.... |
22:44.57 | talntid | but asterisk still will not start, and doesn't really give me any errors to go by |
22:45.20 | talntid | i assume that because those files tried to modify the extensions_*.conf files... that my issue must be there.. |
22:45.37 | radic | hallo WIMPy |
22:45.42 | talntid | is there a way to regenreate those extensions_*.conf files, from command line? the web GUI will not load. |
22:45.46 | WIMPy | Hi radic |
22:46.23 | radic | WIMPy: sag doch das du da bist :P |
22:46.30 | WIMPy | talntid: They are generated by the GUI, so probably not. But you need to ask that in #freepbx. |
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22:56.40 | WIMPy | MFBS! :-( |
22:57.09 | WIMPy | Now my dundi peers will only work when they are cashed. |
22:57.44 | WIMPy | There had to be a catch to the loopback lswitch. It just worked far too well. |
22:58.54 | WIMPy | Is there an way to make a loopback wait for its target? |
23:00.20 | jeffspeff | I'm having troubles with fax detection... I have a fax extension set up on my inbound context, but when the fax comes through it just does an auto fallthrough and doesn't actually detect the fax. does anybody have any ideas? |
23:00.59 | WIMPy | No. I't not about cacheing. It just works sometimes. F*** |
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23:26.48 | ChannelZ | jeffspeff: what version of * |
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23:34.35 | jeffspeff | ChannelZ, 1.8.5 |
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23:41.37 | ChannelZ | hmm. I guess I should test mine and see if it's still working. |
23:42.00 | ChannelZ | There were some problems with prior versions where internally it was trying to jump to a bizarre extension. |
23:42.52 | ChannelZ | You just have faxdetect=incoming set for your channel(s), and an extension called 'fax' in the context for those channels? |
23:43.15 | jeffspeff | ChannelZ, hold on, and i'll do a pastebin of what i have real quick |
23:46.18 | jeffspeff | ChannelZ, this is the relevant parts of extensions.conf ---> http://pastebin.com/PgzyZVGi and then in sip.conf, I have "faxdetect=cng" set under [general], I have also tried "faxdetect=yes", and there wasn't a difference in results. |
23:47.11 | Tim_Toady | jeffspeff: set wait at least to 3 seconds |
23:47.19 | Tim_Toady | takes sometime to detect the tone |
23:47.36 | jeffspeff | Tim_Toady, ok, let me set that and test it again... just a minute |
23:47.47 | Tim_Toady | or maybe 4, its trial and error |
23:48.00 | Tim_Toady | but for me it works at 3-4 |
23:48.35 | jeffspeff | does it hurt to have it set to high? |
23:49.49 | Tim_Toady | no, ur callers will just wait a bit more before reaching u:P |
23:51.00 | jeffspeff | Tim_Toady, ChannelZ, I set the wait for 4 seconds, then sent a fax from our ancient fax machine that's not connected to this system, (asterisk should receive it as an incoming fax), but i got the same results |
23:51.43 | jeffspeff | Tim_Toady, ChannelZ, FYI I'm using fax for asterisk by digium, and trying to configure it to use g.711 |
23:51.47 | ChannelZ | you don't want Wait, you want WaitExten |
23:52.06 | WIMPy | Is there something like a minimum length for dundi requests? It seems to work if I dial more than one digit befor lifting the handset. |
23:52.18 | ChannelZ | oh no, nevermind, fax detect I think happens regardless... |
23:52.27 | WIMPy | Not that anyone would do so... |
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23:54.16 | WIMPy | No. |
23:54.24 | ChannelZ | jeffspeff: going backwards, I don't think "faxdetect=cng" means anything, it should just be 'yes' |
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23:54.45 | WIMPy | makes a big mental not: Always use ! in switch patterns, never .. |
23:56.01 | WIMPy | Let's see what side effects this will have... |
23:56.22 | jeffspeff | ChannelZ, I got that setting from the asterisk book, for version 1.8... you can set faxdetect in sip to be cng, t38, yes, no |
23:56.30 | jeffspeff | but i'll try anything at this point |
23:57.53 | ChannelZ | hmmm.. maybe the/my samples are out of date |
23:58.29 | ChannelZ | Indeed mine are, my bad.. |
23:58.30 | Tim_Toady | jeffspeff: we talk about fax that arrives to asterisk by sip or some fxo port? |
23:58.38 | jeffspeff | sip |
23:58.54 | Tim_Toady | codec set to g711? |
23:59.18 | Tim_Toady | is the call indeed set up in g711? |
23:59.25 | jeffspeff | where do i set that at? i see in my fax license where it has an avaialbe g711 channel |
23:59.53 | jeffspeff | the calls are all using ulaw |
23:59.55 | ChannelZ | IE when you call in with your fax machine, it's not coming in as g729 or something from your provider? |
23:59.58 | Tim_Toady | in sip.conf disallow all other codecs and only allow ulaw |