| 00:01.30 | jsjc | p3nguin: do you see anything that could be wrong here??? I cannot get the bloody problem hehe http://pastebin.com/LZvAzXTE |
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| 00:15.52 | Jed | hey what type of sip phone do most of you guys use? |
| 00:16.37 | WIMPy | make a doodle :-) |
| 00:19.37 | ChannelZ | Linksys SPA922 |
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| 00:52.10 | Jumpie | hey guys |
| 00:52.27 | Jumpie | is there any officially published/documented list of motherboards and other hardware specifically rated to work excellent with asterisk? |
| 00:52.39 | Jumpie | i've gotten unlucky sometimes with particular models...not even cheap ones.just certian ones just have wierd glitchy issues |
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| 00:56.52 | ChannelZ | Not that I've ever seen. The main thing is the PCI bus if you're using TDM cards, a lot of older chipsets might not handle the interrupt pounding that occurs |
| 00:57.48 | ChannelZ | Or not even 'older chipsets', but just certain flakey ones in general. I haven't really heard a lot of complains on any front lately though |
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| 01:10.18 | Jed | Hey, with asterisk, is it possible to merge 2 or more incoming calls into 1 conference call? |
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| 01:11.30 | ChannelZ | MeetMe |
| 01:11.43 | ChannelZ | (so, yes..) |
| 01:12.05 | Jed | can it do an outbound as well |
| 01:14.51 | ChannelZ | I don't understand the question |
| 01:15.39 | Jed | sorry, I know about meetme, but lets say you have your regular extension, you call 1 person, and then someone calls you on the same extension |
| 01:16.01 | Jed | is it possible to combine those 2 calls into 1 call? |
| 01:16.51 | ChannelZ | There's a few ways.. you could transfer everyone into a conference, or some phones can do 3-way themselves actually |
| 01:17.36 | Jed | from what I understand, 3way calling can only be done if you call out |
| 01:17.48 | Jed | or am I wrong about that |
| 01:18.21 | Jed | well if you have 1 person on the line yeah, but in the end that 3rd person would have to be dialed after you have an established connection with the other person |
| 01:18.22 | ChannelZ | Well through a normal POTS line in your house maybe |
| 01:18.23 | WIMPy | Why should initial direction make a difference? |
| 01:18.51 | Jed | initial wouldnt matter |
| 01:19.26 | ChannelZ | In *, at least on my phone, if I am on a call - whether they called me or I called them - and someone else calls, my phone blinks. I can put the person on hold, answer the incoming call, and then merge them together if I want |
| 01:19.45 | Jed | hrm, so its more of a phone function? |
| 01:20.17 | ChannelZ | That is outside of the phone company; That is, I have 4 discreet POTS lines coming into my * box, so the second person is just calling in on another actual line. It's not using the telco's 3-way |
| 01:20.52 | WIMPy | Doing it in the phone has a higher risk of generation echos than doing it in *. |
| 01:21.24 | ChannelZ | To do more than that, I'd have to shove people into a MeetMe room.. but the 3-way is handled by my phone its self (Linksys SPA922) |
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| 01:29.46 | Jed | ok thanks |
| 01:41.28 | jsjc | hoho i think i found about the queue and why did hung up... could be because I had a waitExte in front? |
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| 01:49.54 | fuxu2 | n3glv: you here? |
| 01:56.54 | jsjc | is there anyway to check queue members agents and that stuff from CLI? |
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| 01:58.26 | jsjc | queue show QUEUENAME shows me no members but pretty sure there should be. |
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| 02:03.57 | dieno | hello all |
| 02:04.26 | dieno | can any anyone give me idea how to send "pound key" at the end of extension |
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| 02:04.42 | dieno | like Dial(DAHDI/${EXTEN}# |
| 02:06.42 | dieno | hmmm any one there ? |
| 02:12.53 | hardwire | anybody here interested in a community supported numbering plan? |
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| 02:18.32 | hardwire | sets up some test servers.. time to figure out rt sip peer caching |
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| 02:21.50 | spenguin[work] | test |
| 02:24.03 | hardwire | doesn't really want to reply to that .. but will |
| 02:24.09 | hardwire | 1..2..3..4.. |
| 02:31.13 | spenguin[work] | :> |
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| 02:45.15 | p3nguin | jsjc: WaitExt() isn't a valid application to my knowledge. |
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| 02:52.04 | sat-man | all these people and nobody's talking? |
| 02:54.07 | hardwire | WaitExten() is. |
| 02:55.01 | p3nguin | but WaitExt() is what he's using, so I'm curious if that is causing his failure. |
| 02:56.11 | p3nguin | It is also possible that WaitExt() is valid in a version other than the ones I have used, but I'm not counting on it. |
| 02:58.30 | spenguin[work] | hey, I have an exten defined but when I dial it asterisk shows no sign of it denying the exten but the sip phone just gives up |
| 02:58.39 | spenguin[work] | other extens work |
| 02:58.47 | spenguin[work] | I have verbosity at 9 |
| 02:59.31 | p3nguin | How about a sip debug? |
| 02:59.43 | spenguin[work] | will check |
| 03:00.26 | spenguin[work] | is there anyway I can filter sip debug |
| 03:00.41 | spenguin[work] | theres 100s of sip connections atm :s |
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| 03:02.11 | p3nguin | You can filter by IP address or by peer. |
| 03:02.24 | hardwire | I really need to get on the opensips horse at some point |
| 03:02.30 | hardwire | or sippy |
| 03:02.45 | spenguin[work] | p3nguin: oh cool |
| 03:03.14 | p3nguin | Sometimes the peer doesn't match, so I often prefer to filter by ip. |
| 03:04.00 | spenguin[work] | ok |
| 03:05.45 | p3nguin | What I mean is that sometimes the reason you're wanting to get a sip debug is because the peer doesn't match, so filtering by the peer often doesn't work. |
| 03:06.57 | spenguin[work] | yep, I got debug with sip debug <ip> |
| 03:07.18 | spenguin[work] | but Im not sure where is asterisk denying the exten |
| 03:07.26 | spenguin[work] | SIP/2.0 404 Not Found ? |
| 03:11.08 | spenguin[work] | [Apr 16 08:39:34] Looking for 01730797900 in cos2 (domain 172.16.140.235) |
| 03:11.09 | spenguin[work] | [Apr 16 08:39:34] > |
| 03:11.09 | spenguin[work] | <--- Reliably Transmitting (no NAT) to 172.16.134.25:5060 ---> |
| 03:11.09 | spenguin[work] | SIP/2.0 404 Not Found |
| 03:12.22 | spenguin[work] | ah got it |
| 03:12.45 | spenguin[work] | the context I created wasnt added within the cos domain |
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| 03:33.11 | spenguin[work] | hey I was trying to set callerid with CALLERID(num)=(12344556) |
| 03:33.16 | spenguin[work] | but that wouldnt work |
| 03:33.30 | spenguin[work] | only CALLERID(all)=(12344556) , worked |
| 03:33.31 | spenguin[work] | why |
| 03:45.25 | ChannelZ | well I don't see how either of them worked |
| 03:48.22 | ChannelZ | Set(CALLERID(num)=12345) |
| 03:48.30 | ChannelZ | Set(CALLERID(name)=Throwup) |
| 03:48.49 | spenguin[work] | ChannelZ: well yeah - i didnt bother about the syntax when typing here |
| 03:49.02 | ChannelZ | NoOp(${CALLERID(all)}) results in "Throwup" <12345> |
| 03:49.31 | spenguin[work] | but Set(CALLERID(num)=12345) would work |
| 03:49.39 | spenguin[work] | just Set(CALLERID(all)=12345) |
| 03:49.45 | spenguin[work] | worked |
| 03:50.55 | ChannelZ | yeah and? |
| 03:51.04 | Jumpie | i loved putting an echo test |
| 03:51.07 | Jumpie | for my telemarketer shit script |
| 03:51.24 | Jumpie | make sme wonder how long some carried on a broken conversation with themselves |
| 03:51.26 | ChannelZ | Jumpie: I wish you could put it on a 3 second delay |
| 03:51.49 | Jumpie | there may be a way..just not with teh techo test routine itself |
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| 03:53.22 | spenguin[work] | ChannelZ: so why |
| 03:53.43 | spenguin[work] | on the recieving end Im able to print out using NoOp(${CALLERID(num)}) |
| 03:54.49 | ChannelZ | I don't even know what the hell you're asking anymore |
| 03:55.09 | ChannelZ | You say 'this would work', and 'this worked' so what is the problem |
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| 03:57.53 | p3nguin | Maybe you could record and playback the person rather than just echoing it. |
| 03:58.40 | ChannelZ | I want to instantly forward all spam calls to a call center drone in India |
| 03:59.57 | p3nguin | You could build your own list and do it preemptively or set up some sort of one-touch deal which requires you to at least answer first. |
| 04:01.07 | ChannelZ | waves his hand over the top of his head |
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| 04:13.44 | TJNII | Does anyone use Host Virtual? |
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| 04:25.33 | Jumpie | hey ChannelZ how easy is it to interface software /databases into CID lookup? |
| 04:25.41 | Jumpie | to pull up account screen for example if db matches CID |
| 04:25.48 | Jumpie | is there some kinda utility to help on that? or all manual |
| 04:29.19 | spenguin[work] | ChannelZ: |
| 04:29.30 | spenguin[work] | 02:30 spenguin[work]: hey I was trying to set callerid with CALLERID(num)=(12344556) |
| 04:29.33 | spenguin[work] | 02:31 spenguin[work]: but that wouldnt work |
| 04:29.49 | spenguin[work] | 02:48 spenguin[work]: just Set(CALLERID(all)=12345) |
| 04:29.52 | spenguin[work] | worked |
| 04:30.11 | spenguin[work] | Im wondering why didnt Set(CALLERID(num)=12345), work |
| 04:30.54 | p3nguin | Set(CALLERID(num)=1234567890) or Set(CALLERID(all)=Your Name <1234567890>) |
| 04:31.16 | spenguin[work] | ? |
| 04:31.38 | p3nguin | or Set(CALLERID(name)=Your Name) |
| 04:31.49 | spenguin[work] | Set(CALLERID(name)=xxxxxx), doesnt work either |
| 04:31.50 | p3nguin | Those are the valid ways to set the caller ID info. |
| 04:31.57 | spenguin[work] | probably relatd to some telco issue |
| 04:32.12 | spenguin[work] | p3nguin: I do know that .. |
| 04:37.21 | p3nguin | Does your telco allow you to set either of the values for caller ID? |
| 04:38.20 | spenguin[work] | Set(CALLERID(name)=1234567890) works |
| 04:38.34 | p3nguin | *sigh* |
| 04:38.43 | spenguin[work] | it does |
| 04:38.50 | spenguin[work] | Im not getting what you mean to ask |
| 04:39.33 | spenguin[work] | err scratch that 03:39 spenguin[work]: Set(CALLERID(name)=1234567890) works |
| 04:39.40 | p3nguin | yeah? |
| 04:39.48 | spenguin[work] | Set(CALLERID(all)=1234567890) |
| 04:39.56 | spenguin[work] | ^^ that is the only thing that works |
| 04:40.01 | spenguin[work] | nothing else |
| 04:40.11 | p3nguin | Did you try num instead of all? |
| 04:40.19 | spenguin[work] | did, it doesnt work |
| 04:40.25 | p3nguin | That's strange. |
| 04:40.27 | spenguin[work] | thats what Im trying to ask since a while |
| 04:40.32 | nix8n82 | your probably not using it right |
| 04:40.58 | nix8n82 | a small syntax error |
| 04:41.00 | spenguin[work] | ive got logs - hold on |
| 04:41.16 | p3nguin | core set verbose 9 |
| 04:42.10 | ChannelZ | I think there are two different issues here.. you're trying to set that for the purposes of making a call, seeing what you get on the other side |
| 04:42.36 | ChannelZ | That's a bit between you and your telco. |
| 04:43.55 | spenguin[work] | ChannelZ: yeah Im blaming the telco too, just that on the other side Im able to NoOp(${CALLERID(num)}), and it works |
| 04:46.29 | ChannelZ | When you NoOp it on your side what does it show when you just set (num) ? I'd guess you have some sort of name already in there from the device you're calling from (but maybe not) |
| 04:46.48 | ChannelZ | When you do (all) with just the number, the name gets wiped out. |
| 04:47.00 | spenguin[work] | ill test and post logs |
| 04:50.09 | spenguin[work] | ok |
| 04:50.16 | spenguin[work] | I get what was wrong |
| 04:50.54 | spenguin[work] | when I tested Set(CALLERID(all)=(0123456)) .. that worked |
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| 04:51.07 | spenguin[work] | Set(CALLERID(num)=(0123456)) ... that doesnt |
| 04:51.15 | spenguin[work] | or is not meant to work |
| 04:51.16 | ChannelZ | but that's putting ()s around your number which is not right |
| 04:51.42 | spenguin[work] | cool, I get now |
| 04:52.10 | spenguin[work] | the previous admin has CALLERID(all)= .. all over the dialplan |
| 04:53.11 | p3nguin | Beat him with a blunt instrument. |
| 04:53.26 | spenguin[work] | heh |
| 04:53.31 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 04:53.53 | Micc | This is weird. fax for asterisk on 1.6.2.6 isn't answering. |
| 04:54.28 | ChannelZ | Maybe you didn't pay your faxes today. |
| 04:55.06 | Micc | I put an answer in there before the receivefax, but it just keeps ringing. |
| 04:56.16 | ChannelZ | Hmm that sounds more like a channel problem? What happens if you Answer and then Playback something? |
| 04:56.36 | spenguin[work] | p3nguin: looks like the telco doesnt support name |
| 04:56.42 | p3nguin | typical |
| 04:56.50 | spenguin[work] | ok |
| 04:56.56 | Micc | thats what I'm gonna try next, it must be a problem with the adapter. or maybe this new build of asterisk is screwed up. |
| 04:57.13 | p3nguin | Most US telcos don't support sending the caller ID name because that's not how name display works. |
| 04:58.13 | ChannelZ | Micc: yeah could be either |
| 05:00.16 | Micc | wow, same thing with playback. |
| 05:00.51 | spenguin[work] | p3nguin: ok, Im in India |
| 05:01.12 | Micc | dialin to did works. |
| 05:01.18 | Micc | so it must be the handytone. |
| 05:01.35 | Micc | I'll try a zyxel p2300r-p1 |
| 05:04.00 | Micc | now that I think about it. I had the same problem with another handytone the other day, but I just thought it was my POS analog phone. |
| 05:05.08 | ChannelZ | What does it speak to *? SIP? |
| 05:08.22 | Micc | yeah |
| 05:08.48 | Micc | zyxel is working, but its waiting forever before dialing. |
| 05:09.00 | Micc | I think I need to work on the handytone a bit more. |
| 05:11.48 | Micc | Why does it say licensed channels 2 but max concurrent 1? |
| 05:12.23 | ChannelZ | Why does who say what about what license? |
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| 05:23.42 | Micc | fax show stats says 2 licenses but max concurrent is only 1 |
| 05:23.57 | Micc | I have 1 free fax for asterisk and one real fax for asterisk license. |
| 05:24.05 | Micc | but shouldn't that make 2 concurrent? |
| 05:25.39 | ChannelZ | I think those are statistics.. as in you had 2 concurrent faxes going on at some point in time |
| 05:25.50 | ChannelZ | err 1 I mean |
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| 05:26.11 | p3nguin | fax show licenses |
| 05:26.15 | Micc | oh |
| 05:26.22 | Micc | that makes sense. I think your right. |
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| 05:29.57 | ChannelZ | and now you have 0 concurrent connections. |
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| 06:14.49 | Micc | Why can I send a fax using T.38 but not receive? Its the same adapter. Does SendFax not do T.38? |
| 06:15.17 | Micc | Maybe my adapter only accepts it one way? |
| 06:15.37 | Micc | asterisk doesn't even say its trying to negotiate T.38 though. |
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| 06:46.57 | spenguin[work] | http://laforge.gnumonks.org/papers/gsm_phone-anatomy-latest.pdf |
| 06:47.07 | spenguin[work] | if it interests anyone is pretty damn good read |
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| 07:07.27 | *** join/#asterisk Polysics (~luca@host113-41-static.25-87-b.business.telecomitalia.it) |
| 07:07.30 | Polysics | hello |
| 07:07.40 | Polysics | is there any reason to use IAX2 over SIP? |
| 07:07.45 | Polysics | for my user accounts? |
| 07:09.24 | Polysics | i am reading up, but th epicture is not that clear |
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| 07:16.53 | ChannelZ | It depends. IAX is nice because it goes over 1 port so it can be easier to deal with in NAT/firewall issues. |
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| 07:17.37 | ChannelZ | And it can be trunked so multiple calls can be multiplexed into the same stream |
| 07:21.03 | Polysics | ChannelZ, for my web-based collaboration app, would there be any particular reason to use one over the other? |
| 07:21.19 | Polysics | or if I were to have a better IAX than SIP client, i can go with that? |
| 07:21.59 | viq | There are many more clients for SIP than IAX, and I guess that's about it. |
| 07:23.27 | Polysics | viq, there aren't many good browser-embeddable clients for either technology, knowing that i can use either allows for slightly more freedom on the client side |
| 07:23.42 | ChannelZ | At the end of the day, whatever works is whatever works |
| 07:25.06 | ChannelZ | IAX should be more effecient with trunking but that doesn't matter in your case. |
| 07:25.23 | ChannelZ | Although I guess it's supposed to be more effecient in general |
| 07:25.24 | viq | *shudder* browser clients... |
| 07:25.54 | viq | But I see what you're going for |
| 07:26.10 | Polysics | viq, i would rather use XMPP with voice extension, tbh |
| 07:26.18 | viq | yeah |
| 07:26.29 | viq | have you seen http://sip-communicator.org/ ? |
| 07:26.31 | Polysics | but SIP/IAX allows for a wider range of applications |
| 07:26.47 | Polysics | no but i will be happy to look at it .-) |
| 07:27.22 | viq | I don't know much besides "java sip client", but maybe that makes it browser-embeddable - or not. |
| 07:27.27 | Polysics | sip communicator is basically what i am building as a web-based, closed app :-) |
| 07:27.37 | Polysics | i suppose you need a "proper" applet, tbh |
| 07:27.55 | Polysics | i am not a Java expert, and almost all I know relates to JSP programming |
| 07:28.20 | viq | That's more than me ;P |
| 07:28.24 | Polysics | i am a PHP expert, and a Ruby and C/C++ user, which basically is useless in the VoIP field, apparently :-) |
| 07:28.41 | Polysics | clients are either written in C#/VB.NET or Java |
| 07:30.42 | ChannelZ | Why must this be a webapp? |
| 07:33.14 | viq | Polysics: http://www.doddling.com/ |
| 07:33.15 | Polysics | ChannelZ, long story short, it is supposed to not require a "true" installation |
| 07:33.39 | Polysics | besides, a web app does have its own merits, they just lie OUTSIDE the VoIP issue :-) |
| 07:34.23 | Polysics | doddle looks interesting |
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| 08:20.48 | *** join/#asterisk SayKyo (~SayKyo@212.232.17.120) |
| 08:20.57 | SayKyo | hi everyone |
| 08:21.58 | SayKyo | I need advice with configuring chan_ss7 on an asterisk box in a bit of a special scenarion, can someone help me please? |
| 08:22.06 | SayKyo | *scenario |
| 08:24.42 | SayKyo | I need to make an ss7 connection to two different PC's but with a signalling link only on one of the links |
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| 08:40.56 | Jumpie | tryin to run a mini telco are we? |
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| 08:41.33 | SayKyo | something like that :D |
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| 09:11.56 | SayKyo | how do I tell chanss7 that an ss7 signalling endpoint is reachable via another ss7 point code (do ss7 routing) ? |
| 09:12.15 | SayKyo | do I do this at the driver level ? |
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| 09:49.11 | J4zen | anyone well-known with the AMI command 'Originate'? I'm trying to have it call an external number via SIP-trunk then play an audio fragment using Festival. The command looks like this : originate sip/102 application festival "this is a testcall" |
| 09:49.23 | J4zen | now in stead of dialing extension 102, i want it to call an external number |
| 09:49.35 | J4zen | any clue how to do so? |
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| 09:57.29 | jmkgreen | I've upgraded from 1.4 to 1.6 and now AMI commands to originate calls do absolutely nothing at all. I get "manager.c: Manager received command 'Originate'" in the logs together with all the other passed data, then nothing happens. |
| 09:59.20 | J4zen | jmkgreen: sorry, i don't run any 1.6 versions where i actively use originate so i wouldn't know. |
| 10:00.05 | J4zen | jmkgreen: but since we're on the subject, i posted a question several minutes ago to which you might know the answer |
| 10:00.24 | jmkgreen | It's working fine under 1.4, but under 1.6 it just exits immediately, almost as though the dialplan no longer matches |
| 10:00.40 | J4zen | jmkgreen: using the AMI command 'Originate'? I'm trying to have it call an external number via SIP-trunk then play an audio fragment using Festival. The command looks like this : originate sip/102 application festival "this is a testcall" |
| 10:01.06 | J4zen | jmkgreen: you got verbosity set to a high level?/ |
| 10:01.11 | jmkgreen | 1000 |
| 10:01.22 | J4zen | that'll do. |
| 10:01.39 | J4zen | paste your cli input |
| 10:02.01 | jmkgreen | J4zen: We originate calls and if that succeeds we pass control to an AGI to control the call |
| 10:03.01 | jmkgreen | http://pastebin.ca/1863087 |
| 10:03.08 | jmkgreen | that's from the verbose log |
| 10:03.36 | jmkgreen | as you can see, it accepts an originate command and obtains all the variables, but does nothing afterwards and the connection is effectively dropped |
| 10:04.13 | jmkgreen | In the dialplan we have NoOp()s to track things but they don't even fire |
| 10:25.03 | *** join/#asterisk Dovid (~annon@213.8.121.90) |
| 10:25.06 | Dovid | j #asterisk-il |
| 10:25.21 | *** join/#asterisk jozu (~linuz@84.120.220.193.dyn.user.ono.com) |
| 10:25.25 | jozu | hi to all |
| 10:25.28 | Dovid | hi. anyone here use asterisk Moh+ Realtime ? |
| 10:25.32 | Dovid | hello jozu |
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| 11:15.07 | *** mode/#asterisk [+o file] by ChanServ |
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| 11:53.57 | jsjc | I wonder if I have agents set on agents.conf |
| 11:54.19 | jsjc | that does not mean they are logged into any queue, so I can log them in and off as I require right? |
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| 13:02.58 | slashtom | anyone here know gnugk well? i am trying to route calls from h323 clients connected to gnugk into Asterisk (oh323) |
| 13:03.12 | *** join/#asterisk Dalboni (~Dalboni@189-84-160-249.sodobrasil.net.br) |
| 13:05.07 | Dalboni | Hello everyone. Somebody here uses 1.4 with queue_log patch to write in the DB? It works ok? |
| 13:05.16 | *** join/#asterisk SuPrSluG (~SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 13:08.25 | *** join/#asterisk Polysics (~luca@host113-41-static.25-87-b.business.telecomitalia.it) |
| 13:08.27 | Polysics | hello |
| 13:08.54 | Polysics | would the heavy presence of uber-firewalled potential users skew the decision towards IAX instead of SIP? |
| 13:09.12 | Polysics | IAX2 looks more manageable and explainable to a firewall admin |
| 13:09.34 | Polysics | or would a user sitting behind closed ports necessarily need a sort of STUN/whatever server? |
| 13:09.59 | Polysics | i have this user sitting behind a locked-in firewall that is still able to use Skype - is it due to Skype using port 80 as a fallback? |
| 13:10.36 | Katty | morning |
| 13:12.02 | [TK]D-Fender | Polysics: IIRC there are many Cisco phones capable of VPN connections... I suppose those would be best |
| 13:12.19 | Polysics | [TK]D-Fender, stuck with web-based phones again :-( |
| 13:13.03 | leifmadsen | Polysics: STUN often helps in certain situations. If Asterisk is configured correctly though, rarely have I needed to enable anything for a client behind a firewall |
| 13:13.16 | [TK]D-Fender | Polysics: NO COMMENT |
| 13:13.56 | Polysics | leifmadsen, does IAX present any advantage there? other than being easier to explain to a firewall admin |
| 13:14.53 | Polysics | by the way, leifmadsen, thanks for the zoiper pointer - it is good enough, though not great |
| 13:15.14 | leifmadsen | what does the firewall admin have to do with anything? |
| 13:15.20 | leifmadsen | I don't make ANY changes to my firewalls |
| 13:15.29 | leifmadsen | i.e. I don't forward ports to my phones |
| 13:16.49 | Polysics | some public government administrations have firewall closed from the inside, ie. a user can't connect TO an outside port |
| 13:17.07 | Polysics | and before [TK]D-Fender says it, yes, i get to work with all sorts of idiots :-) |
| 13:17.19 | leifmadsen | gotcha, well it's not easier to explain -- there is nothing more to explain than that you need more than 1 port for SIP |
| 13:17.51 | leifmadsen | IAX2 == 1 port for media and signalling. SIP == 1 port for signalling, multiple ports for media |
| 13:17.55 | Polysics | is there any way to obviate to such a situation? |
| 13:18.14 | Polysics | ie. users are behind a firewall that block outgoing connections to "special" ports too |
| 13:18.19 | leifmadsen | no |
| 13:18.24 | leifmadsen | SIP uses multiple ports for media |
| 13:18.26 | leifmadsen | that's all there is to it |
| 13:18.31 | leifmadsen | IAX2 only uses 4569 |
| 13:18.49 | leifmadsen | I don't understand "obviate to such a situation" |
| 13:19.20 | Polysics | user is behind a firewall that blocks outgoing connections save for the few "standard" ports - is there a solution that does not involve opening ports? |
| 13:19.43 | leifmadsen | how do you expect to communicate without communicating? |
| 13:19.55 | *** join/#asterisk ZeeekBot (~anonymous@pdpc/supporter/active/zeeek) |
| 13:20.00 | leifmadsen | if you can't make an outgoing connection... you can't talk to the remote end |
| 13:20.00 | ZeeekBot | heh |
| 13:20.02 | Polysics | standard = 80, 443, 110, 143, 25 |
| 13:20.16 | Polysics | what you say makes a lot of sense .-) |
| 13:20.26 | leifmadsen | then you need an asterisk server that will listen for IAX2 on one of those ports |
| 13:20.32 | Polysics | i wonder what happens if i set Asterisk to bind on port 80 |
| 13:20.36 | leifmadsen | you'll never support very many SIP calls with that limited space |
| 13:20.57 | leifmadsen | remember that the MEDIA of SIP is actually RTP, which is a dynamic range of ports |
| 13:21.06 | leifmadsen | each call requires about 2 ports per call |
| 13:21.14 | file | and it's UDP, not TCP... so if they only allow TCP then no soup for you |
| 13:21.21 | leifmadsen | that too :) |
| 13:21.28 | leifmadsen | file: HI! |
| 13:21.36 | file | leifmadsen: NO! |
| 13:21.40 | leifmadsen | bitch |
| 13:21.51 | leifmadsen | <3 |
| 13:21.57 | ZeeekBot | Katty: is it hot in here or is it just me? |
| 13:22.15 | ZeeekBot | "this is the last call before we close the file" |
| 13:22.26 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 15:19.34 | *** join/#asterisk infobot (ibot@rikers.org) |
| 15:19.34 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.6, 1.6.1.18, 1.6.0.26 (2010/03/12), 1.4.30 (2010/03/12), *-Addons 1.6.2.1, 1.6.1.3, 1.6.0.5, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 15:20.40 | *** join/#asterisk jasonwert-work (~jasonwert@99-27-170-70.lightspeed.cicril.sbcglobal.net) |
| 15:21.21 | leifmadsen | TJ^: I'm sure someone does since it was invented |
| 15:21.55 | mdg | TJ^: I came across this while reading Ch. 8 --> http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-8-SECT-4.html |
| 15:22.00 | mdg | shrugs |
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| 15:27.49 | Zeeek | doing the splits |
| 15:28.08 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 15:28.25 | Zeeek | We're starting the VUC in 10-15 minutes, jump on #vuc if you're interested |
| 15:29.26 | ChannelZ | which is what |
| 15:29.42 | TJ^ | leifmadsen mdg thanks |
| 15:29.52 | TJ^ | leifmadsen the guy who invented is a nutter |
| 15:30.02 | TJ^ | mdg u implemented any QoS? |
| 15:30.04 | leifmadsen | his mutters nutter? |
| 15:30.24 | TJ^ | nutter = crazy |
| 15:30.34 | [TK]D-Fender | nutters mutter |
| 15:30.52 | TJ^ | is confused... mutter? |
| 15:30.53 | TJ^ | anyway |
| 15:30.57 | mdg | TJ^: no |
| 15:31.08 | TJ^ | so i dont suppose u know anything bout DSCP? |
| 15:31.17 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.97) |
| 15:31.43 | ChannelZ | Now I want some Nutter Butters. |
| 15:32.47 | [TK]D-Fender | PEANUT BUTTER JELLY TIME |
| 15:33.06 | Zeeek | why is [TK]D-Fender never auto tab completable? |
| 15:33.18 | Zeeek | I haven't slept since i discovered this |
| 15:33.27 | TJ^ | [TK]D-Fender yes it is |
| 15:33.37 | TJ^ | [TK]D-Fender [TK]D-Fender [TK]D-Fender works fine here |
| 15:33.51 | Zeeek | [TK]D- |
| 15:33.55 | Zeeek | mine won't |
| 15:34.28 | Zeeek | actually, any name with a - will not complete properly on this client |
| 15:34.33 | Zeeek | but I wonder why? |
| 15:34.44 | TJ^ | which client? |
| 15:35.21 | Zeeek | this happens to be Colloquy but it seems to me I had the problem elsewhere too |
| 15:35.38 | Zeeek | wish I had some peanut butter heer |
| 15:35.49 | Naikrovek | don't use some weird obscure client and you'll have better results i'm betting |
| 15:36.09 | Naikrovek | oh you're on a mac |
| 15:36.12 | Naikrovek | ALL BETS ARE OFF |
| 15:36.42 | Zeeek | I know that's easy to say, but it's the same on other clients too. Like the iPod :) |
| 15:36.59 | Naikrovek | apple = inferior w/ superiority complex |
| 15:36.59 | leifmadsen | I like xchat |
| 15:37.12 | Zeeek | I tried xchat and it blew something up |
| 15:37.18 | Zeeek | so I gave up |
| 15:37.35 | Naikrovek | you're in a walled garden |
| 15:37.35 | Zeeek | but I have never given up on the VUC |
| 15:37.54 | Zeeek | Naikrovek: no I'm not, I have a Windows XP box next to the Mac! |
| 15:38.03 | Naikrovek | treasure it |
| 15:38.13 | Zeeek | is getting ready to be as funny on #vuc starting in a few minutes |
| 15:38.32 | Zeeek | COme one come all to the only real taste of VoIP COmedy |
| 15:38.49 | Zeeek | It's like if Jeremy McNamara and BKW were hosting SNL |
| 15:39.02 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 15:39.06 | Zeeek | ~Seen JerJer |
| 15:39.07 | infobot | jerjer <~PhatJ@asterisk/original-h323-guy/JerJer> was last seen on IRC in channel #asterisk, 33d 11h 9m 16s ago, saying: 'anyone know of any US based PSTN test numbers? i'm looking to test q.931 disconnect causes... unallocated number and what not'. |
| 15:39.25 | Zeeek | ~seen bkw |
| 15:39.26 | infobot | bkw <~bachman@84.243.3.42> was last seen on IRC in channel #debian, 1d 32m 51s ago, saying: 'Doesn't iscsi target handle to export a block device such as Path=/dev/md0{p1,p2,p3}? My initiators can mount successful, but no drive appear on the initiator. But when I use Path=/dev/md0 the initiator can mount and the drive appear.'. |
| 15:39.43 | Zeeek | ~seen yermutha |
| 15:39.44 | infobot | i haven't seen 'yermutha', Zeeek |
| 15:39.54 | Zeeek | ok, if you do tell her I'm in #vuc |
| 15:41.01 | coppice | ~seen nees t'nevah i |
| 15:41.01 | infobot | i haven't seen 'nees t'nevah i', coppice |
| 15:41.44 | Zeeek | that went way, way over my head but I sense an inherent comic genius in there hidden behind the obscurity |
| 15:43.48 | *** join/#asterisk lep (~leo@93-62-167-170.ip23.fastwebnet.it) |
| 15:44.41 | Zeeek | last chance: http://vuc.me or just jump on #vuc, you're half way there |
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| 15:52.56 | LemensTS | http://pastebin.ca/1863421 does anyone have a clue why it is putting my AGI variables into array[1] and not seperating them out? ive tried commas and | |
| 15:54.56 | Qwell | LemensTS: try something released this year? |
| 15:55.32 | [TK]D-Fender | LemensTS: because args aren't separated by COMMAS, they are separated by SPACES |
| 15:55.56 | [TK]D-Fender | <PROTECTED> |
| 15:56.29 | Qwell | 2009-01-21 |
| 15:57.15 | LemensTS | TK: exten => s,1,AGI(callblast2-program.php|${PHID} ${CAMPAIGNID} ${NUMBER} .... ? |
| 15:57.39 | LemensTS | Ive never seen or done it that way |
| 15:58.23 | LemensTS | Qwell: ok ill upgrade it if i cant figure anything else out |
| 15:59.01 | Qwell | why waste your time trying something over a year old? |
| 15:59.10 | Qwell | why waste ours by asking before upgrading? |
| 16:01.17 | *** part/#asterisk lep (~leo@93-62-167-170.ip23.fastwebnet.it) |
| 16:01.32 | LemensTS | I dont think spaces is right. |
| 16:02.50 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 16:03.02 | *** join/#asterisk jhirley (~jhirley@mail.mmdlaw.com) |
| 16:03.10 | LemensTS | Qwell: its on some server in pakistan, that has other live telephone programs running on. Clients system. Not sure what else they are doing, upgrading this while making sure everything else remains working while dealing with different timezone is a last resort...looks like thats where im at...thanks for the advice |
| 16:03.16 | leifmadsen | Qwell: do you have a place I can upload an 11MB file? |
| 16:03.31 | Qwell | leifmadsen: depends on the reason for it O.o |
| 16:03.35 | leifmadsen | Qwell: you'll like it |
| 16:03.47 | Qwell | could try sending through jabber |
| 16:03.54 | leifmadsen | sure send me a msg |
| 16:06.35 | *** join/#asterisk DMeloUK (~DominicMe@64.129.93.210) |
| 16:10.03 | *** part/#asterisk wunderkin (~kbockman@pool-71-106-236-25.lsanca.dsl-w.verizon.net) |
| 16:13.50 | bmoraca_work | here's a fun little problem... |
| 16:13.57 | bmoraca_work | i get no audio when i use local channels |
| 16:15.02 | bmoraca_work | has anyone experienced any kind of similar issue? |
| 16:16.31 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 16:30.05 | leifmadsen | bmoraca_work: sounds like an open issue I've been looking at in the bug tracker |
| 16:30.29 | bmoraca_work | do you have a ticket number i could look at to see if it's the same? |
| 16:30.35 | leifmadsen | M17151 |
| 16:30.38 | MuffinMan | [acknowledged] [Asterisk] Applications/app_playback 0017151: Playback deadlock? reported by RobinFood https://issues.asterisk.org/view.php?id=17151 |
| 16:30.40 | leifmadsen | the topic is misleading |
| 16:31.10 | bmoraca_work | doesn't seem to be the same to me |
| 16:31.12 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.134.90.dsl.dyn.forthnet.gr) |
| 16:31.13 | leifmadsen | ok |
| 16:31.22 | bmoraca_work | i get no audio at all |
| 16:31.25 | bmoraca_work | no ringing, nothing |
| 16:32.41 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.77) |
| 16:35.05 | *** join/#asterisk williamD (~william@173-12-140-201-jacksonville.hfc.comcastbusiness.net) |
| 16:37.20 | [TK]D-Fender | bmoraca_work: show us the dial command |
| 16:37.41 | williamD | Need some info on the setup of the zapata.conf... We have 2 PRI and would like to use channels 1-15 and 25-41 for incoming calls only and 16-23 and 42-47 for outbound only...... |
| 16:38.00 | bmoraca_work | [TK]D-Fender: exten => *567,1,Dial(Local/*98@from-internal) |
| 16:38.04 | bmoraca_work | actually |
| 16:38.06 | bmoraca_work | i take that back |
| 16:38.08 | bmoraca_work | that one worked |
| 16:38.16 | bmoraca_work | playback works over local channels |
| 16:38.20 | bmoraca_work | dialing out a SIP trunk doesn't |
| 16:38.21 | p3nguin | williamd: What's the problem? |
| 16:38.27 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:38.35 | [TK]D-Fender | bmoraca_work: use "/n" to prevent the channel from re-bridging |
| 16:38.48 | bmoraca_work | hmmmmm |
| 16:39.04 | williamD | how do i set it up so that incoming calls hit onlt the channels 1-15 and 25-41?? |
| 16:39.19 | williamD | onlt=only |
| 16:39.24 | *** join/#asterisk Mango (~iMango@d154-20-86-138.bchsia.telus.net) |
| 16:39.25 | *** part/#asterisk DMeloUK (~DominicMe@64.129.93.210) |
| 16:40.13 | Mango | I'm trying to set up DNS SRV so that my Linksys devices will use a backup Asterisk server if the primary is unreachable. |
| 16:40.51 | Mango | When I set Proxy Redundancy Method to Normal, the behavior is very erratic. The phone (SPA-921) doesn't respect priority and ping-pongs between the servers. |
| 16:41.12 | [TK]D-Fender | williamD: You don't control what channels gets used. The telco does. Ask them to change their selection |
| 16:41.19 | bmoraca_work | [TK]D-Fender: no go, unfortunately |
| 16:41.31 | Mango | When I set Proxy Redundancy Method to Based on SRV Port, things get a little better. The phone uses the primary server until I knock it offline and then switches to the backup server. But, it never switches back. |
| 16:41.35 | [TK]D-Fender | bmoraca_work: And it works with reinvites disabled when dialed directly? |
| 16:41.36 | Mango | I have Proxy Fallback Intvl set to 30. |
| 16:41.43 | Kobaz | williamD: why do you need specific channels? |
| 16:41.54 | bmoraca_work | [TK]D-Fender: reinvites enabled or disabled, it works when dialed directly |
| 16:42.02 | [TK]D-Fender | bmoraca_work: Also do a test with the primary leg for * internal indications. You may have a fault indications.conf |
| 16:42.13 | williamD | I have and they told me that we need to sned them a soft busy on the channels in the outbound group.... |
| 16:42.37 | [TK]D-Fender | williamD: Ugly, and they are retarded |
| 16:42.47 | [TK]D-Fender | williamD: So get coding... "Congestion()" |
| 16:43.05 | p3nguin | That doesn't seem like the best way to do it, but if they won't cooperate, what other options do you have? |
| 16:43.18 | Kobaz | williamD: why do you need this? |
| 16:43.22 | *** join/#asterisk rservers (~brian@unaffiliated/rservers) |
| 16:44.20 | williamD | we are getting alot of collisions of in and outbound calls |
| 16:44.29 | Kobaz | your telco is retartded |
| 16:44.34 | Kobaz | retarded |
| 16:44.48 | p3nguin | retar dead |
| 16:44.50 | Kobaz | tell them to fix their call handling |
| 16:45.43 | Kobaz | and what do you mean by collisions |
| 16:46.12 | williamD | inbound and out bound calls hit at the same time....... |
| 16:46.32 | Kobaz | that still makes no sense |
| 16:46.42 | williamD | they say they cannot set up trunk groups on PRI that way |
| 16:46.55 | Kobaz | do you mean an outbound call will try and use the same channel that an inbound call is already using? |
| 16:47.01 | williamD | YES |
| 16:47.33 | Kobaz | there's something wrong with the channel negotiations then |
| 16:47.46 | *** join/#asterisk kartik (~koolkarti@117.199.115.173) |
| 16:48.11 | Kobaz | either your line card drivers are screwy, or your telco is screwy |
| 16:49.39 | williamD | this only happens during peak hours.... |
| 16:50.27 | Kobaz | so what happens on a "collision" |
| 16:50.45 | williamD | when it happens you get 90% of the time a fast busy and hang up and 10% calls and conected... |
| 16:51.13 | Kobaz | and how you do you know it's picking the same channel as an established call |
| 16:52.22 | Kobaz | unless you can prove that it's a driver issue... it sounds like your telco is running out of capacity |
| 16:52.50 | williamD | out bound caller and inbound call can talk..... seems that the incoming and outgoing calls try to grab the line at the same time |
| 16:53.02 | Kobaz | show us some debugging to prove it |
| 16:54.32 | Kobaz | have you watched pri intense debug? |
| 16:54.55 | williamD | i will turn on log , are busy time is 8-10am EST |
| 16:55.29 | williamD | i have tried its hard to keep up when you have 40+ calls |
| 16:56.33 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 16:56.35 | wcselby | o/ |
| 16:57.32 | Kobaz | write it out to a log file |
| 16:57.52 | williamD | this is a custom system that Didium built when they had custome dev tem in place.... |
| 16:58.06 | williamD | team |
| 16:58.09 | *** part/#asterisk LemensTS (~LemensTS@adsl-70-238-175-27.dsl.stlsmo.sbcglobal.net) |
| 16:58.22 | williamD | I wish they were still around....... |
| 17:00.14 | *** join/#asterisk mv2 (~maverick@83.240.229.38) |
| 17:00.51 | mv2 | is the dtmf end(end bit on) packet needed to allow asterisk to detect dtmf in rfc mode ? |
| 17:01.16 | *** join/#asterisk nix8n82 (~chatzilla@63.162.27.14) |
| 17:01.34 | TJ^ | anyone know ANYTHING about QoS/diffserv for asterisk? |
| 17:01.47 | Naikrovek | QoS, yes |
| 17:01.51 | Naikrovek | never heard of diffserv |
| 17:01.58 | TJ^ | ah crap |
| 17:02.04 | TJ^ | Tos? CoS? |
| 17:02.07 | TJ^ | DSCP? |
| 17:02.25 | Naikrovek | what? |
| 17:02.36 | TJ^ | how dyu do ur QoS? |
| 17:02.41 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 17:02.51 | Naikrovek | TJ^: my router does outgoing QoS |
| 17:03.01 | TJ^ | 0o what router u got? |
| 17:04.15 | Naikrovek | cisco 2811 |
| 17:04.53 | TJ^ | dyu know what layer it is? |
| 17:04.58 | TJ^ | how u setup the QoS on it? |
| 17:05.04 | *** join/#asterisk MAbbas (~abbas@203.215.177.194) |
| 17:05.05 | Naikrovek | routers are L3 devices |
| 17:05.05 | TJ^ | how u setup the QoS on asterisk? |
| 17:06.03 | TJ^ | did u basically just tell the router prioritise the SIP protocol or the PBX host or specific packets? |
| 17:06.05 | MAbbas | Hi All, how can I transfer a call from agent1 to agent2? |
| 17:07.52 | wcselby | any recommendations on dahdi echo cancellers? |
| 17:08.03 | wcselby | mg2, kb1, sec2? |
| 17:08.12 | Qwell | hpec |
| 17:08.36 | wcselby | is that compiled in by default? |
| 17:08.45 | *** join/#asterisk Z_God (~julius@schwartzenberg.xs4all.nl) |
| 17:08.45 | MAbbas | My calls are coming to Queue(), how can agent1 transfer call to agent2? |
| 17:10.13 | Qwell | wcselby: No. It's also not free |
| 17:10.24 | wcselby | Qwell - that's what I thought |
| 17:10.25 | leifmadsen | MAbbas: do a transfer on the phone |
| 17:10.26 | kerframil | wcselby: hpec (commercial) or oslec. don't bother with the others; they're nigh-on useless. |
| 17:10.37 | mv2 | is the dtmf end(end bit on) packet needed to allow asterisk to detect dtmf in rfc mode ? |
| 17:10.47 | wcselby | kerframil - gotcha. i was afraid of that. |
| 17:10.58 | leifmadsen | mv2: I want to say, "No, asterisk doesn't care too much", but i'm not 100% sure on that |
| 17:11.15 | wcselby | do dahdi changes in 1.4.x require asterisk to be restarted before they're picked up? |
| 17:11.25 | MAbbas | <@leifmadsen> : I tried that, but it logs out agent1.. |
| 17:11.30 | leifmadsen | o.O |
| 17:11.33 | leifmadsen | then you have something setup wrong |
| 17:12.15 | MAbbas | <@leifmadsen> : What do you mean by "transfer on phone"? u mean transfer to agent2's extension? |
| 17:12.21 | leifmadsen | yes |
| 17:12.33 | leifmadsen | I mean hit the transfer button, or use a built-in transfer |
| 17:12.44 | leifmadsen | if your agent is being logged out, then you have something configured wrong |
| 17:12.55 | leifmadsen | or at least configured so that is the expected behaviour |
| 17:13.03 | MAbbas | I tried that but, it logs out agent1 |
| 17:13.22 | leifmadsen | well I have no data to determine why that is the case |
| 17:15.06 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 17:15.22 | MAbbas | <@leifmadsen> can u help me with dialplan then? |
| 17:15.42 | leifmadsen | MAbbas: I can't right now -- just ask the room generally. You're going to want to show what you're getting at the console in a pastebin |
| 17:15.46 | bmoraca_work | urg |
| 17:15.48 | p3nguin | That's so annoying to see leifmadsen being quoted with something he hasn't said. |
| 17:15.50 | leifmadsen | if I can respond, I will |
| 17:15.50 | bmoraca_work | figured out the root of my issue |
| 17:15.57 | bmoraca_work | how to figure out a solution to the problem |
| 17:15.57 | leifmadsen | p3nguin: eh? |
| 17:16.01 | MAbbas | ok.. |
| 17:16.08 | Kobaz | hey |
| 17:16.08 | leifmadsen | p3nguin: oh ya... his tab completion is weird :) |
| 17:16.14 | leifmadsen | is for horses |
| 17:16.26 | p3nguin | leifmadsen: I bet you he's copying/pasting every time he wants to address someone. |
| 17:16.36 | leifmadsen | giggles a bit |
| 17:16.49 | p3nguin | I've encountered that several times. |
| 17:17.27 | Kobaz | leifmadsen: you know that lib_pri/dahdi supports calls coming from a line configured as att.... even though the asterisk box is set up as NI2 |
| 17:17.29 | p3nguin | mabbas: When you want to address a person, type a few letters of the person's nick and press the tab key to complete it. Press it more than once if there is more than one match. |
| 17:17.39 | leifmadsen | Kobaz: I didn't know that :) |
| 17:17.47 | leifmadsen | Kobaz: I try to do as little hardware as possible |
| 17:17.50 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 17:18.41 | *** join/#asterisk afink (~afink@204.26.87.226) |
| 17:19.13 | MAbbas | My simplest of dialplan for this is: exten => _80X,1,Queue(testQ) exten => _XXX,1,AgentLogin(${EXTEN}) exten => _XXXX,1,Dial(SIP/${EXTEN}) |
| 17:19.51 | p3nguin | Looks okay to me. |
| 17:19.57 | Kobaz | leifmadsen: heh |
| 17:20.27 | Kobaz | leifmadsen: i think we're dumping sangoma cards... we switched to adtran sip gateways for this one install |
| 17:20.49 | leifmadsen | Kobaz: nice, I wanted to try one of those |
| 17:20.53 | Katty | peeks in |
| 17:20.54 | leifmadsen | PRIs into the box, then SIP out to Asterisk? |
| 17:21.11 | Kobaz | leifmadsen: it's soooooo great... tech support is great |
| 17:21.11 | Kobaz | yeah |
| 17:21.16 | leifmadsen | Kobaz: good to know |
| 17:21.28 | Kobaz | because... i can't call up sangoma and say... why is dahdi crashing?? |
| 17:21.34 | Kobaz | they dont care about asterisk |
| 17:21.38 | leifmadsen | agreed |
| 17:21.41 | Kobaz | their card is working fine... that's the end of their support |
| 17:21.49 | leifmadsen | they love that asterisk exists for their hardware, but that's about all |
| 17:21.55 | Kobaz | yeah |
| 17:22.18 | Kobaz | and it's about the same price per port as sangoma/etc |
| 17:22.29 | Kobaz | ~720 per t1 |
| 17:22.49 | [TK]D-Fender | Kobaz: Really? what models? |
| 17:23.01 | Kobaz | [TK]D-Fender: TA904 |
| 17:23.14 | Kobaz | echo cancelled and everytning |
| 17:23.34 | Kobaz | ssh/telnet/web interface |
| 17:23.44 | Kobaz | auto email alerts, snmp exports |
| 17:24.04 | [TK]D-Fender | Kobaz: Wow... excellent value |
| 17:24.08 | Kobaz | i know |
| 17:24.20 | Kobaz | they make a quad span for $2k, but only two ports do pri |
| 17:24.29 | Kobaz | they are coming out with the netvanta 600 series Q2 though |
| 17:24.43 | Kobaz | which will have a single span, dual span, quad spad, octal span versions |
| 17:25.05 | Kobaz | so by summer there will be affordable external 8 span pri gateways |
| 17:25.24 | *** join/#asterisk QaDeS_ (~mklaus@p4FC70894.dip0.t-ipconnect.de) |
| 17:25.27 | Kobaz | [TK]D-Fender: the best thing about it, is i get the hardware out of the asterisk box |
| 17:25.35 | Kobaz | [TK]D-Fender: i'm really tired of fighting with dahdi/libpri crashes |
| 17:26.33 | Kobaz | [TK]D-Fender: 10 year warrenty too |
| 17:29.28 | p3nguin | So you'll be going SIP only on Asterisk after the change? |
| 17:29.30 | leifmadsen | Kobaz: and much easier to fail over between asterisk boxes when you're only doing SIP |
| 17:29.40 | leifmadsen | to me that's the best way of doing it |
| 17:29.50 | leifmadsen | keep the PRI stuff separate of your call handling |
| 17:31.17 | Kobaz | leifmadsen: exactly |
| 17:31.30 | Kobaz | p3nguin: yeah |
| 17:31.36 | Kobaz | p3nguin: it makes stuff soooo much simpler |
| 17:31.53 | Kobaz | previously i was looking at audiocodes and stuff |
| 17:32.05 | Kobaz | and it's like shit man... i dont want to spend $3000 for a single t1 gateway |
| 17:32.41 | Kobaz | p3nguin: and it has grounding support, so if you get some surges in the line it'll take care of that |
| 17:32.42 | p3nguin | Got an image of that $750 Adtran? |
| 17:32.51 | Kobaz | p3nguin: search for TA904 |
| 17:33.13 | p3nguin | Nice looking. |
| 17:34.13 | p3nguin | So you can go straight from the smart jack to the TA904, then Ethernet from the TA904 to Asterisk? |
| 17:34.17 | *** join/#asterisk jasonwert-work_ (~jasonwert@99-27-170-70.lightspeed.cicril.sbcglobal.net) |
| 17:34.37 | Katty | hi |
| 17:34.44 | Kobaz | okay here's the thing |
| 17:34.52 | Kobaz | i've been told they aren't certified to go directly to the pstn |
| 17:35.00 | Kobaz | so you can throw in a $100 csu and then go |
| 17:35.14 | Kobaz | but tech support said they have customers who do go straight to a smart jack and have no problems |
| 17:36.33 | Kobaz | they don't have a built in csu, but really they aren't expensive to add on |
| 17:37.00 | p3nguin | So it is technically capable, but just not certified. I wonder if the telco ever complains about them being hooked direct. |
| 17:37.01 | Kobaz | p3nguin: but yeah, smart jack to 904, ethernet to asteisk, and done |
| 17:37.17 | Kobaz | dont have to power down the asterisk |
| 17:37.41 | Kobaz | i wish sangoma made gateways |
| 17:37.54 | Kobaz | they kind of already have 90% of one |
| 17:38.03 | Kobaz | they have the sip stack for the A10X's |
| 17:38.15 | Kobaz | they jsut need to put it in a box |
| 17:38.43 | *** join/#asterisk SaintJean (~xyz@64.229.75.107) |
| 17:40.14 | p3nguin | and that thing has four FXS ports, too. |
| 17:40.18 | Kobaz | yeap |
| 17:40.21 | Kobaz | some of them have fxos too |
| 17:40.25 | Kobaz | if you look at like the 908 |
| 17:40.40 | Kobaz | p3nguin: it took me about 2 days to learn how to configure them |
| 17:40.46 | Kobaz | if you know cisco it should go even quicker |
| 17:41.05 | Kobaz | although you can do almost everything from the web interface |
| 17:41.20 | SaintJean | Hi, how should I make load balancing with my .CallFile over SIP line? Thanks |
| 17:41.36 | p3nguin | It has a built-in DSL modem, too? |
| 17:41.42 | Kobaz | depends which one you get |
| 17:41.49 | Kobaz | there's a t1 version and a dsl version |
| 17:42.48 | Kobaz | you can use it as a t1 router too |
| 17:42.59 | p3nguin | I was looking at the specs on provantage.com ... they don't seem to distinguish those features. |
| 17:43.03 | Kobaz | like, say you get a ppp t1 from your telco with data/voice |
| 17:43.12 | Kobaz | you can split off the voice to fxs/pri |
| 17:43.16 | Kobaz | and route the data over ethernet |
| 17:43.23 | Kobaz | or split off the voice to sip |
| 17:43.32 | p3nguin | They make it seem like it has capability of T1, Ethernet, E&M, DSL... all in the same model. |
| 17:43.35 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 17:43.41 | Kobaz | no, it's seperate models |
| 17:43.52 | Kobaz | look at the adtran product pages |
| 17:43.55 | p3nguin | k |
| 17:49.01 | wcselby | Kobaz - did you look at the redfone offerings? I've always been curious about those? |
| 17:49.10 | Kobaz | i haven't really |
| 17:49.43 | p3nguin | I'm liking the looks of these TA products. I never knew what they had to offer. |
| 17:50.12 | Kobaz | looks like it's not straight sip |
| 17:50.24 | Kobaz | some sort of propriatary stack on the pbx side |
| 17:50.40 | Kobaz | p3nguin: yeah so far i like them, one box in production, and four more on the way |
| 17:51.30 | Kobaz | i have them sitting in the lab |
| 17:51.32 | Kobaz | need to set them up |
| 17:52.25 | p3nguin | I take it there is nothing else of similar quality with like features. I'm assuming you've done your homework to decide on the 904s. |
| 17:53.11 | Kobaz | yeah |
| 17:53.20 | Kobaz | i looked at like 6 different gateways |
| 17:58.59 | *** join/#asterisk R-Guy (daemon@mony.mcleodnet.com) |
| 17:59.47 | Kobaz | how much does a 3000 foot spool of cat3 run |
| 18:01.31 | bmoraca_work | cat3? |
| 18:01.36 | bmoraca_work | why would you use cat3? |
| 18:01.50 | Jed | use with an ata? |
| 18:02.11 | bmoraca_work | cat3 is usually the same price and some times more expensive than cat5 |
| 18:02.40 | p3nguin | Are you looking for 25 pairs or what? |
| 18:03.03 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 18:03.05 | bmoraca_work | Kobaz: also, you can't go wrong with those Adtran TA900s. They're freaking wonderful. |
| 18:03.27 | Kobaz | oh, fonebridge does have regular isdn->sip |
| 18:03.31 | Kobaz | but adtran has been around longer |
| 18:03.36 | Kobaz | bmoraca_work: yeah |
| 18:04.50 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 18:05.02 | bmoraca_work | they work great as media gateways between T1 and SIP. and they can go either way. |
| 18:05.09 | Kobaz | yeap |
| 18:05.18 | bmoraca_work | the firewall/router portion of them isn't terrible either |
| 18:06.12 | p3nguin | How many pairs are you wanting in your Cat 3 cable? |
| 18:06.32 | Kobaz | enough to do dsl |
| 18:09.31 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 18:09.48 | *** part/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 18:11.19 | *** join/#asterisk rare1980_ (~rare1980@12.25.228.67) |
| 18:11.22 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:13.27 | p3nguin | I can't find Cat 3 in more than 1000 ft spools. |
| 18:13.47 | Katty | call Cape Electric |
| 18:14.04 | p3nguin | Cat 3, 6 conductor round, 1000 ft is $76. |
| 18:14.19 | p3nguin | 4 conductor flat, 1000 ft is $58. |
| 18:14.52 | p3nguin | If you need 3000 ft on a single spool, my supplier isn't going to come through on this deal. |
| 18:15.58 | Kobaz | mm |
| 18:16.22 | Kobaz | twist some wires together at the ends? |
| 18:16.26 | Kobaz | heh |
| 18:16.28 | p3nguin | haha |
| 18:18.07 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:18.13 | Kobaz | still trying to think about this 1/2 mile network run |
| 18:18.24 | Kobaz | cant do wireless. no los at all |
| 18:18.32 | Kobaz | i was thinking of t1 or dsl |
| 18:18.32 | p3nguin | fiber |
| 18:18.45 | Kobaz | there are poles between drops |
| 18:18.59 | Kobaz | burying would be... rather fun |
| 18:19.01 | Kobaz | lots of rocks |
| 18:19.53 | Kobaz | i was thinking something on the less expensive end... fiber gets up there |
| 18:20.26 | *** join/#asterisk Lyma (~Lyma@unaffiliated/lyma) |
| 18:20.41 | p3nguin | Copper would require additional hardware since the signal doesn't travel that far on copper normally. |
| 18:21.15 | Lyma | hi ppl! |
| 18:22.12 | bmoraca_work | stupid asterisk and stupid PIX 525 |
| 18:22.22 | Kobaz | p3nguin: yeah |
| 18:22.37 | *** join/#asterisk SuGaO (~SuGaO@189.89.145.254) |
| 18:22.41 | bmoraca_work | in order to fix my local channel-no audio problem, i had to answer the call, play a blank audio file to establish the RTP translation, and then dial my local channel |
| 18:22.48 | bmoraca_work | it works, but it's hokey! |
| 18:23.25 | *** join/#asterisk LeddyHM (leddy@you.cant.hack.thisbox.org) |
| 18:24.12 | *** join/#asterisk Knightfal (~j@mailer.1callres.com) |
| 18:24.13 | Kobaz | p3nguin: a t1 csu can bump up to 6000 feet or something... can't it? |
| 18:24.37 | LeddyHM | Looking for some guidance.. users can no longer call our main number, just dead silence. people can call DID, and our fax # maintained by the same system.. sip provider is saying they are receiving "proxy authentication" and telling us to fix our pbx |
| 18:24.46 | LeddyHM | everything worked yesterday and no changes have been made |
| 18:24.59 | LeddyHM | any ideas where I can go? I think it's the dang provider |
| 18:25.24 | SuGaO | Hi! |
| 18:25.32 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:26.13 | Lyma | hi SuGaO. |
| 18:26.23 | Lyma | u r wellcom |
| 18:26.42 | R-Guy | Kobaz: there are 900MHz wireless that do not require LOS to go half mile. |
| 18:28.00 | Kobaz | oh |
| 18:28.06 | Kobaz | R-Guy: what power levels? |
| 18:28.25 | R-Guy | radiated power around 100mW |
| 18:28.35 | Kobaz | mmm |
| 18:29.11 | Kobaz | http://www.trangobroadband.com/wireless-products/multipoint-broadband-access/M900S-900mhz.aspx |
| 18:29.12 | R-Guy | The ones we used gave us around 2Mbps thoughput. |
| 18:29.15 | Lyma | anyone with this error: "ERROR[7360]: chan_dahdi.c:8803 dahdi_pri_error: !! Got reject for frame 75, but we have nothing -- resetting!" ? |
| 18:30.06 | R-Guy | Also got around 25km with LOS and 9 element yagis on each end. |
| 18:30.08 | Lyma | using dahdi 2.3.0 + 2.3.0 + asterisk 1.4.30 |
| 18:30.16 | *** join/#asterisk ChkDigit (~mike@static24-72-71-175.r.rev.accesscomm.ca) |
| 18:30.25 | Kobaz | nifty |
| 18:31.49 | *** join/#asterisk freezey (~trees@static-64-61-84-174.isp.broadviewnet.net) |
| 18:32.56 | freezey | odd issue... so from internal i have 1 extension that uses follow me and internal it works fine i can dial the ext and it goes to the external number... from external i can call the direct dial # and it fwds me to the correct follow me number as well... but from another site i have hooked up through dundi it fails and gives a 503 error any ideas? |
| 18:33.04 | R-Guy | It would probably be cheaper (and much easier) than stringing cable arial or doing direct burial. |
| 18:34.12 | Kobaz | R-Guy: yeah |
| 18:39.47 | bmoraca_work | lots of problems with voice over wireless, though |
| 18:39.53 | bmoraca_work | retransmits kill you |
| 18:39.58 | bmoraca_work | quality suffers |
| 18:40.57 | Lyma | jitter too... |
| 18:41.03 | bmoraca_work | yes |
| 18:41.07 | bmoraca_work | lots of jitter on wireless |
| 18:41.09 | Chainsaw | Yes, WMM doesn't help as much as they say. |
| 18:41.24 | Chainsaw | Going 802.11A would probably help. |
| 18:41.31 | bmoraca_work | depends |
| 18:41.35 | bmoraca_work | on the environment |
| 18:41.38 | Chainsaw | (The modulation schemes are less complicated and the frequency spectrum is less congested) |
| 18:42.27 | Chainsaw | So an interference spike will blow away less data. 802.11A has fatter channels. (802.11N stole that idea as "fat channels" for 2.4GHz; but 802.11A had it first) |
| 18:46.21 | LeddyHM | any ideas on "proxy authentication denied" for 1 number as mentioned above? |
| 18:47.41 | LeddyHM | err "proxy authentication required" |
| 18:48.04 | *** join/#asterisk Zagrophyte (~TRosenbau@fw.swerdlin.net) |
| 18:48.30 | Zagrophyte | Anyone willing to answer a TAPI question, or is there a better channel for that? |
| 18:48.33 | p3nguin | kobaz: ANSI T1.403 seems to indicate that a T1 CSU port maximum distance is 6200 feet (1890 meters). This doesn't seem to correspond with the normal rule of 655 feet without needing a repeater, though. So I'm not real sure. |
| 18:50.05 | [TK]D-Fender | LeddyHM: confirm the perr that it is matching and make sure that its set to "insecure=port,invite" |
| 18:50.09 | [TK]D-Fender | peer* |
| 18:51.51 | *** join/#asterisk Igramul (~black@p5497F2A4.dip.t-dialin.net) |
| 18:52.13 | R-Guy | bmoraca_work: I'm talking about a point-to-point wireless bridge -- no retransmits, very little jitter. |
| 18:52.35 | Zagrophyte | Is there anything about TAPI I should be aware of before building a central service app that monitors multiple lines via TAPI for a telephony integration project I'm working on? |
| 18:54.04 | LeddyHM | yeah I'm looking through that now |
| 18:54.18 | LeddyHM | everything looks same but still analyzing |
| 18:57.59 | p3nguin | kobaz: I found something else similar on a CSU/DSU product sheet: DSX-1 mode: 655 ft. (199.6 m); CSU mode: 6000 ft. (1828.8 m) using 24 AWG cable. |
| 18:58.05 | vader-- | have you guys ever heard of a PY-40 IP Phone? |
| 18:58.32 | vader-- | PY-90 |
| 18:59.38 | vader-- | http://www.chinaroby.com/english/py-90-ip%20phone.htm |
| 19:00.40 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 19:02.14 | *** join/#asterisk SSJGotenks (~41ffe7e3@gateway/web/freenode/x-gdfdfwpddtvgbvzy) |
| 19:02.41 | Kobaz | p3nguin: nifty |
| 19:02.53 | Kobaz | p3nguin: that's kinda what i thought i remember reading |
| 19:02.57 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:04.04 | p3nguin | kobaz: And the T-Extender's spec shows 5000 feet between repeaters. |
| 19:04.11 | [TK]D-Fender | Chinaroby has a long history of spamming the VoIP-info wiki... cheap chinese crap |
| 19:05.28 | [TK]D-Fender | And this on in particular looks like a knockoff of the equally shitty PA1688 ones done by atcom |
| 19:06.05 | p3nguin | "If you have a T1 CSU that is limited to 655 feet, connected to a device limited to 655 feet, and you put the T-Extender at one end, the T-Extender will drive the extra distance. However, the far end device will still be limited to 655 feet and will not likely have sufficient "drive" to reach the far end. The T1 transmitted signal is the issue addressed by the T-Extender, so in this case, the line would be error-prone or possibly only ... |
| 19:06.05 | p3nguin | ... work in one direction." |
| 19:06.11 | p3nguin | So for a long run, they say use a T-Extender at both ends. |
| 19:08.02 | *** part/#asterisk Zagrophyte (~TRosenbau@fw.swerdlin.net) |
| 19:09.08 | *** join/#asterisk KingDavidNYC (~Chris1232@rrcs-69-193-218-18.nyc.biz.rr.com) |
| 19:09.20 | KingDavidNYC | hello |
| 19:11.45 | LeddyHM | hrm, our original crappy vendor configured a username and password on 2 of the trunks |
| 19:15.26 | *** join/#asterisk Alagar (~Administr@122.164.35.5) |
| 19:17.49 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:17.55 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 19:21.06 | vader-- | what is a decent cheap but feature rich and quality IP phone for a small business? |
| 19:21.23 | leifmadsen | heh |
| 19:21.31 | leifmadsen | Polycom 3xx series |
| 19:21.43 | leifmadsen | ~cheap |
| 19:21.44 | infobot | from memory, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 19:21.45 | Kobaz | i love polycoms. i wish they had more buttons |
| 19:22.07 | Jed | doesnt seem like any one is a fan of ciscos here |
| 19:22.12 | afink | that is some good advice. I bought cheap phones and now we upgrading all of them to polycom |
| 19:22.36 | afink | we got snoms which have lots of features and buttons but the voice quality sucks |
| 19:23.15 | afink | Some cheaper linksys ones I bought are doing pretty good |
| 19:23.33 | Kobaz | linksys phones seem to be decent |
| 19:23.38 | Kobaz | aastra makes good sounding phones too |
| 19:24.06 | Kobaz | Jed: cisco requires a license per-phone, even after you buy them |
| 19:24.12 | Kobaz | Jed: and i think it's yearly too |
| 19:24.25 | Jed | even for sip? |
| 19:24.28 | Kobaz | yeah |
| 19:24.35 | vader-- | so 331 or 335 would be good for a small business |
| 19:24.41 | vader-- | that needs like 4-5 phones |
| 19:25.03 | *** join/#asterisk blaines (~blaines@wsip-24-248-37-18.ph.ph.cox.net) |
| 19:25.43 | Kobaz | 331s are nice, with the second ethernet port |
| 19:25.57 | *** join/#asterisk dajhorn (~dajhorn@adsl-75-17-124-26.dsl.rcsntx.sbcglobal.net) |
| 19:26.12 | [TK]D-Fender | vader--: Planning on PoE? Need a passthrough? Whats your lighting like? |
| 19:26.58 | vader-- | not sure i have to ask my friend |
| 19:27.01 | vader-- | im trying to help him out |
| 19:27.12 | Kobaz | [TK]D-Fender, hah... i thought you said lightning |
| 19:27.30 | Kobaz | my lightning is rather... charged up |
| 19:28.30 | *** join/#asterisk coppice (~chatzilla@202.64.176.93) |
| 19:28.49 | [TK]D-Fender | Kobaz: Looks like I really need to brush up on Adtran's product line |
| 19:29.30 | Kobaz | [TK]D-Fender: they make some really nice stuff |
| 19:29.36 | ChannelZ | I use Linksys 922s. I'm don't know that I'd call myself "a fan" but they've worked great for me and I like them well enough |
| 19:30.01 | [TK]D-Fender | ChannelZ: I recommend Linksys in places where Everyone else gets hit with nasty import premiums. |
| 19:30.14 | R-Guy | No annual license for Cisco phones running SIP. One-time fee for phones running SCCP. |
| 19:30.17 | WIMPy | has a 962 and I haven't even wired it up any more. |
| 19:30.31 | Kobaz | [TK]D-Fender: they are coming out with the netvanta 600 series in Q2 2010.. 1x/2x/4x/8x pri to sip boxen... right now they just have the 904s, which are 2xt1 and 4xt1... but they can only do 1xpri and 2xpri respectively |
| 19:31.22 | Kobaz | R-Guy: i remember reading about some weird licensing requirements for sip setups |
| 19:31.24 | [TK]D-Fender | vader--: forget the 331 and go right to 335. so in order of precedence: 321, 335, 450, 650. |
| 19:31.54 | [TK]D-Fender | vader--: 5XX = worthless, 430,33X, 320, 30X = bleh value |
| 19:32.08 | [TK]D-Fender | (If you're buying new at current standard retail) |
| 19:32.18 | R-Guy | As soon as the Call Manager supported SIP users (staring with 5.0), the model changed from licensing the phone to licensing the call manager. |
| 19:32.49 | KingDavidNYC | Hello, does anyone knows of a dialplan function that returns how many people are left in a meetme room |
| 19:33.19 | Kobaz | R-Guy: i worked at a place that used crisco phones... i don't know if it was configuration, or this is just how it worked... but if a phone lost connection you had to reboot the entire switch since it couldn't reconnect |
| 19:33.56 | R-Guy | We run Call Manager and that is not the case here. |
| 19:34.15 | *** join/#asterisk moos3 (~rgenthner@216.52.121.66) |
| 19:34.29 | Kobaz | and noone had any login to the box, so that was our only fix |
| 19:35.13 | moos3 | I hate asterisk directory thats built in there their a replacement any where? some how out the the key presses 374 its not finding anything that begins with a F |
| 19:35.21 | moos3 | but finding simpson |
| 19:35.27 | moos3 | which makes no sense |
| 19:36.04 | *** join/#asterisk zomenox (~brock@254120.firewall.winthrop.edu) |
| 19:36.41 | keith4 | I just set up res snmp. Is enterprises.digium all that's available? (.1.3.6.1.4.1.22736) I was hoping to see SIP peers and such in here |
| 19:45.52 | wcselby | moos3 - does your simpson match have a first name that starts with an "F"? |
| 19:47.15 | zomenox | Hi, i have a question i cant seem to find an answer for |
| 19:47.30 | [TK]D-Fender | moos3: Would help if you showed us something to debug... |
| 19:47.31 | zomenox | I have an asterisk box, using ODBC |
| 19:47.47 | Qwell | he needs to upgrade |
| 19:47.47 | zomenox | for the extensions file |
| 19:47.56 | zomenox | this works great |
| 19:48.02 | zomenox | but wildcards do not work |
| 19:48.04 | Qwell | ~enter |
| 19:48.05 | infobot | the enter key is not a substitute for punctuation. Use a period '.', exclamation mark '!', question mark '?', comma ',', colon ':', semi-colon ';' emdash '--', or ellipsis '...' instead. |
| 19:48.20 | *** join/#asterisk moos3 (~rgenthner@216.52.121.66) |
| 19:48.29 | Qwell | moos3: what version of Asterisk? |
| 19:48.32 | moos3 | wcselby, no |
| 19:48.47 | moos3 | 1.6.0.15 |
| 19:48.47 | wcselby | moos3 - can you show us some cli from when this happens? |
| 19:49.00 | Qwell | moos3: upgrade |
| 19:49.12 | wcselby | moos3 - along with the relevant name in voicemail.conf? |
| 19:49.22 | wcselby | moos3 - or you could upgrade like Qwell is asking |
| 19:49.42 | moos3 | si the newer version going to fix this? |
| 19:50.23 | [TK]D-Fender | moos3: Fix what? We don't see a problem... |
| 19:52.05 | moos3 | i'm trying to get the key press to show up in the console |
| 19:52.12 | *** join/#asterisk blaines (~blaines@93.sub-75-242-233.myvzw.com) |
| 19:52.22 | moos3 | that way i can paste them |
| 19:52.32 | wcselby | core set verbose 10 |
| 19:54.29 | moos3 | i'll have to debug, need to run to the bank, and then will post |
| 19:55.41 | *** join/#asterisk roe_ (~admin@smtp.harmonytownshipschool.org) |
| 19:56.03 | Qwell | I really don't understand why people waste their (and our) time by testing against such old versions. |
| 19:56.24 | roe_ | Is it possible to get qualify output via SNMP in version 1.4? I would like to graph response time |
| 19:56.53 | coppice | Qwell: don't you know that the earliest version of any software is sure to be the best |
| 19:57.01 | *** join/#asterisk ChkDigit (~mike@static24-72-71-175.r.rev.accesscomm.ca) |
| 19:57.32 | KingDavidNYC | Hi, does anyone knows of a way to get the number of people left on a conference room from the dialplan? |
| 20:01.21 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 20:04.22 | [TK]D-Fender | KingDavidNYC: core show functions <- |
| 20:04.30 | [TK]D-Fender | KingDavidNYC: If you don't see it.. then the answer is "no" |
| 20:07.13 | KingDavidNYC | no, it ant't there.... too bad, it sounds like a must have function |
| 20:07.42 | KingDavidNYC | TK-Fender: Thanks |
| 20:08.42 | *** join/#asterisk MindTheGap (~caio@187.20.150.193) |
| 20:09.43 | WIMPy | agi it yourself |
| 20:10.47 | KingDavidNYC | WIMPy: you mean like send an AMI with <meetme list confno> and then parse the output? |
| 20:11.27 | WIMPy | e.g. |
| 20:11.51 | KingDavidNYC | WIMPy: I guess I will have to :( |
| 20:12.02 | MindTheGap | hello ppl, what are you using as a switchboard for asterisk 1.6? i know fop and fop2 and i dont like them, im mean, i tried Asterboard a while ago and it is much more sleek but it was only for 1.4 so now im trying to find it again and zakotel does not exist anymore. any good free ones out there? |
| 20:12.23 | wcselby | Qwell - not everyone can upgrade to the latest and greatest version? |
| 20:12.37 | Qwell | wcselby: they're doing something wrong then |
| 20:12.41 | WIMPy | Hmmm |
| 20:13.42 | roe_ | not to stick my head in the lion's mouth, but I believe migrating from 1.4 to 1.6 requires a libc6 upgrade, which for production servers, is non-trivial |
| 20:13.51 | WIMPy | Actually, when you use macros, the macro name would show as current conext in AGI, wouldn't it? That way you could implement rather complex new function by just calling an empty macro and do the work |
| 20:13.55 | WIMPy | via AMI. |
| 20:14.39 | *** join/#asterisk bn-7bc (bjarne@2001:470:dc32:0:21b:63ff:fec0:de99) |
| 20:14.40 | wcselby | roe_ - i think qwll was referring to moving to the latest release of 1.6.0.x for the particular case we were talking about |
| 20:14.54 | roe_ | oh |
| 20:14.56 | wcselby | roe_ - but he could have meant the guy needed to go to 1.6.2.x, i dunno |
| 20:15.05 | bmoraca_work | MindTheGap: try iSymphony |
| 20:16.57 | *** join/#asterisk brezular (~brezular@bband-dyn222.178-40-64.t-com.sk) |
| 20:18.41 | MindTheGap | bmoraca_work, last time i saw it was windows only... any good web based one? there is monast but im looking at more options. |
| 20:19.00 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 20:19.38 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:19.46 | leifmadsen | sounds like it might be time to develop your own |
| 20:19.51 | Micc | I'm trying to send a fax with T.38 which worked last night from my home fax machine. But now with the same setup except for a different fax, it dials and the fax says there was no fax answer, but I can hear it pickup and beep. |
| 20:20.21 | leifmadsen | ok, my brain hurts, so I'm done for the weekend... lates |
| 20:20.28 | Micc | I thought the adapter was suppose to still answer with the standard fax squeeling, then interpret that into T.38 |
| 20:21.23 | *** join/#asterisk Alagar (~Administr@122.164.35.5) |
| 20:21.40 | Micc | I setup the adapter as T.38 passthrough, maybe its suppose to be g711, then it'll do T.38 on the other side. |
| 20:22.54 | Micc | I'm gonna try that. |
| 20:24.27 | *** join/#asterisk dwayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 20:24.48 | [TK]D-Fender | checkout time, later all |
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| 20:52.31 | mdg | peace |
| 20:52.32 | *** part/#asterisk mdg (~mdg@unaffiliated/mgroman) |
| 20:54.26 | *** join/#asterisk Pan3D (~Pan3D@streaming.sensoryresearch.net) |
| 20:55.16 | ChannelZ | love |
| 20:55.28 | ChannelZ | and chocolate donuts |
| 20:55.46 | Lyma | and pizza! |
| 21:07.40 | Corydon76-dig | Would that be chocolate frosted donuts or glazed chocolate donuts? |
| 21:08.55 | ChannelZ | I'll take either |
| 21:12.12 | *** join/#asterisk \malex\ (Dmyu3vL9@unaffiliated/malex/x-000000001) |
| 21:14.02 | ChrisWi | prefers jam jamed sugared donuts without hole :) |
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| 21:25.44 | *** join/#asterisk QaDeS_ (~mklaus@p4FC70894.dip0.t-ipconnect.de) |
| 21:27.12 | \malex\ | does anyone have any experience hooking up an allworx 9112 phone to an asterisk box? google doesn't seem to know about such things |
| 21:29.39 | ChannelZ | Google is such a fat slag |
| 21:31.56 | \malex\ | i know! and it doesnt' wash its hands after it goes potty |
| 21:32.00 | *** join/#asterisk rare1980_ (~rare1980@115.186.14.224) |
| 21:39.04 | Qwell | \malex\: Is it a SIP phone? |
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| 21:58.15 | *** join/#asterisk grandpapadot (~nonya@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 21:58.38 | grandpapadot | Hey guys, anyone have a good anonymous/unknown caller id rejection script/logic? |
| 22:01.53 | ChannelZ | define really good? |
| 22:02.38 | ChannelZ | I have a dopey little thing I wrote in PHP that uses a couple of MySQL tables to do CID translation and set variables based on numbers I blacklist |
| 22:03.03 | *** join/#asterisk jsjc (~chatzilla@115.131.202.81) |
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| 22:18.01 | rue_house | . |
| 22:18.12 | ChannelZ | ! |
| 22:18.27 | p3nguin | .--. ..- -. -.-. - ..- .- - .. --- -. |
| 22:19.34 | ChannelZ | You want to suck your what? |
| 22:19.37 | rue_house | so, so far the only problems with the polycom phones have been bad audio, no tech support, takes up half the deskspace, no tiltable screen, dosnt show you dialed digits, and no CSA approval |
| 22:20.04 | rue_house | or UL canada |
| 22:20.21 | *** join/#asterisk smooth_penguin (~smoove@59.95.16.202) |
| 22:20.24 | p3nguin | Other than that, you're satisfied with them? |
| 22:20.41 | ChannelZ | I'll take 10 |
| 22:20.50 | rue_house | oh no I missed a lack of programmable buttons |
| 22:21.25 | rue_house | and of course they cost more than aastra |
| 22:22.23 | rue_house | but its not like you can talk to a polycom sales rep about it |
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| 22:34.50 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.35.181) |
| 22:44.44 | [TK]D-Fender | rue_house: You're about the only one reporting bad audio (We know your analog card is a serious source of this), Tech support? Don't know anyone who has really needed this. Takes up a lot of space? Pots of people love the big screen and the functionality the real-estate adds. And there plenty of compact models to choose from as well. How many quality phone have a tiltable screen? No... |
| 22:44.46 | [TK]D-Fender | ...to Aastra, Linksys, and Cisco.... |
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| 23:27.01 | *** join/#asterisk fnordus (~dnall@70.70.0.215) |
| 23:27.30 | pkecastillo | Hey guys, I'm working with the command originate, and I wonder if you can place an application + an extension?, For example, originate SIP / exten @ context playback application tt-monkeys |
| 23:27.35 | pkecastillo | (and then check) extension exten @ context |
| 23:27.39 | pkecastillo | SIP / exten @ context tt-monkeys playback application extension exten @ context |
| 23:28.03 | pkecastillo | is this possible? |
| 23:28.55 | [TK]D-Fender | pkecastillo: You sepcify either a single Application OR an Extension. Not both |
| 23:29.52 | pkecastillo | hummç |
| 23:30.02 | pkecastillo | ~[TK]D-Fender~ thanks!! |
| 23:35.12 | p3nguin | pkecastillo: originate SIP/yourphone application playback tt-monkeys |
| 23:35.44 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 23:35.50 | p3nguin | pkecastillo: originate SIP/yourphone extension 18005551212@outbound_calls |
| 23:37.17 | pkecastillo | yes, I had that clear, but needed to launch these two options. You know how to send through AMI? |
| 23:38.14 | pkecastillo | ~p3nguin~ because, as I told you I need to send these two orders...., thanks for your help! |
| 23:38.18 | pkecastillo | ;D |
| 23:38.49 | p3nguin | I don't understand. |
| 23:39.10 | [TK]D-Fender | pkecastillo: You don't need to do both. Extension execute APPLICATION. so if you want to do multiple.. have multiple priorities in your target extension. |
| 23:45.29 | pkecastillo | ~[TK]D-Fender~ is that I do not want to put it in a .conf |
| 23:45.30 | infobot | pkecastillo: okay |
| 23:45.36 | pkecastillo | directly send it via command line. |
| 23:45.40 | pkecastillo | or AMI |
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| 23:50.32 | p3nguin | oh no. |
| 23:51.11 | pkecastillo | in the line of exten => s,1,Playback(tt-monkeys) for example. But i need change frequently the playback name, and then send call. I do not write it in plain text is used. |
| 23:54.12 | p3nguin | Originate it like I showed you. |
| 23:55.20 | *** join/#asterisk dwayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 23:56.46 | [TK]D-Fender | pkecastillo: what will determing your multiple instructions if not in a dialplan? |
| 23:57.13 | [TK]D-Fender | pkecastillo: if its only a file to play back, etc you can set VARIABLES etc for the Originated call. |
| 23:57.35 | [TK]D-Fender | pkecastillo: this works for AMI originate & call files, not CLI originate |
| 23:59.23 | *** join/#asterisk DerkKo (~afernande@75-149-178-131-Miami.hfc.comcastbusiness.net) |
| 23:59.31 | pkecastillo | I need, with the same channel call the phone number and playback and redirect a DID number different!, you can showme how i do this?.. |
| 23:59.55 | pkecastillo | ~[TK]D-Fender~ thanks for you help! ;) |
| 23:59.58 | DerkKo | Im wondering how mixmonitor works. |