| 00:00.22 | miamiseb | dahdi_tool also tells you |
| 00:01.24 | ChannelZ | Hmmm. I actually don't get anything in /proc/dahdi |
| 00:02.09 | ChannelZ | something odd is going on here.. hmm. |
| 00:04.06 | *** join/#asterisk mattozzao (~mattozzao@201.76.250.175) |
| 00:05.05 | ChannelZ | huh this is odd, now I'm getting "/usr/share/dahdi/waitfor_xpds: Missing astribank_is_starting" yet it claims to be successfully starting dahdi_dummy afterwards |
| 00:05.48 | miamiseb | Out of sync version of dahdi and dahdi_tool is likely |
| 00:05.51 | ChannelZ | Ah-HA. I think I'm missing libusb |
| 00:05.57 | miamiseb | unfortunately, I have to dip. |
| 00:06.05 | ChannelZ | I'm using the 'complete' DAHDI 2.3.0 |
| 00:07.41 | miamiseb | If your getting high usage via one method you may try switching it to another available method rtc->high precision timer or vice versa |
| 00:07.59 | miamiseb | don't forget to check the actually accuracy of your timing source and make sure its good though |
| 00:08.24 | ChannelZ | yah thanks for the kick |
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| 00:18.05 | wwalker | from AMI Originate |
| 00:18.13 | wwalker | ignore that |
| 00:21.17 | ChannelZ | ok this is bizzarre. Now it's not bukilding/trying to build dahdi_dummy. WTF |
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| 00:25.07 | ChannelZ | huh. 2.2.1.1 does build it. This is something screwy with 2.3.0 |
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| 00:39.27 | arthurh | What's the best way to inject an arbitrary audio stream into bridged channels at arbitrary times? |
| 00:40.13 | arthurh | I've been looking at app_jack.. just wanted to make sure I'm looking in the right direction |
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| 00:41.07 | Tim_Toady | ChannelZ dahdi_dunny is no longer included, if you want to build it edit the makefile by hand |
| 00:41.25 | Tim_Toady | i think its in the readme or install file |
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| 00:59.01 | ChannelZ | Ah. I see they added "core timing" which 'essentially moves the function of dahdi_dummy into the core of DAHDI'. |
| 00:59.18 | ChannelZ | That makes a little more sense I guess, but suffice to say, it's working poorly compared to dummy. |
| 01:00.07 | ChannelZ | oh.. actually even THAT isn't on by default apparently. |
| 01:00.16 | Tim_Toady | res_timing_pthread.so is not suitabl for u? |
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| 01:02.28 | Tim_Toady | anway if you need dahdi_dummy just rebuild dahi-linux, you just have to uncomment one line in drivers/dahdi/Kbuild |
| 01:02.33 | Tim_Toady | obj-$(DAHDI_BUILD_ALL)$(CONFIG_DAHDI_DUMMY)+= dahdi_dummy.o |
| 01:04.05 | ChannelZ | hmm.. I'm going to have to play with this more later.. now it sounds like when you include DAHDI 2.3.0, there are 3 different ways for timing? 1. 'core timing' turned on in a .h file; 2. dahdi_dummy turned back on; 3. pthread which is something I didn't even know about until just now |
| 01:05.11 | devmod | has anyone been able to playback the demo files off aupix website? (h.263) www.aupix.com/index.htm?/support/astvideo.htm |
| 01:05.14 | Tim_Toady | <PROTECTED> |
| 01:05.26 | ChannelZ | right |
| 01:05.37 | ChannelZ | I just meant as a whole |
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| 02:22.03 | TJNII | Apparently Big O tires uses Asterisk. |
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| 02:25.02 | Barometer | anyone have an idea how much a system would cost for a SMB with about 10-15 employees for a system? |
| 02:26.28 | [TK]D-Fender | Barometer: about 100$ for a used PC, and whataver you'd use for phones |
| 02:27.06 | [TK]D-Fender | Barometer: about $120 ea lets say |
| 02:27.50 | ManxPower | TJNII, Asterisk voice work is only a small part of Allison's career, she does PBX's, animation, commercials, etc |
| 02:29.54 | KavanS | ManxPower, I for one, think Allison is attractive |
| 02:30.01 | TJNII | ManxPower: Yea, but those recordings sound awfully familiar. |
| 02:30.08 | TJNII | Eh, who knows. |
| 02:39.56 | p3nguin | She is the IVR voice, after all. |
| 02:40.21 | Jumpie | are there 'total voice replacement' packages for asterisk? |
| 02:40.28 | Jumpie | for some reason, i get a lot of requests for female british |
| 02:40.52 | [TK]D-Fender | Jumpie: Many. Go hit the WIKI |
| 02:41.20 | Jumpie | k ty |
| 02:41.30 | p3nguin | Report what you find so I can check it, too. :) |
| 02:42.31 | *** join/#asterisk drfreeze (~Jim@207.191.114.82) |
| 02:42.39 | drfreeze | Hey, Anyone have a problem with the message light going on, on a Polycom Phone, when it doesn't have VM defined? |
| 02:42.47 | Jumpie | i had thought the voicepacks were just a few items, didnt know you could do whole change out |
| 02:42.50 | Jumpie | ye aim checkin up on it now |
| 02:43.06 | drfreeze | I'm guessing there may be a sip notify way to turn it off |
| 02:43.30 | TJNII | Jumpie: There is a full, drop in British language pack for Asterisk. I forget where I found it, I think asterisk.org |
| 02:44.04 | p3nguin | jumpie: http://www.enicomms.com/cutglassivr/ |
| 02:44.07 | TJNII | I installed it for no particular reason, eliciting many "WTF is the hash key" comments from my friends. |
| 02:44.17 | Jumpie | haha |
| 02:46.43 | [TK]D-Fender | To the left of the pot key.... |
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| 02:56.42 | drfreeze | [TK]D-Fender: Do you know how to turn off MWI on Polycom phone? |
| 02:57.07 | drfreeze | I have a few phones where the MWI light seems to 'randomly' come on |
| 02:58.45 | ChannelZ | are they asking for MWI of someone else's mailbox or somethin? |
| 02:58.56 | drfreeze | Or better yet, why it would light up if the phone has no voicemail |
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| 02:59.52 | drfreeze | ChannelZ: this ext has no mailbox line in sip.conf, it is not in voicemail.conf and there is not ext directory in voicemail |
| 03:00.46 | drfreeze | I used to have problems getting MWI to work, now I can't turn it off. ;) |
| 03:01.06 | TJNII | You fixed it too well. |
| 03:01.28 | drfreeze | :) |
| 03:02.05 | ChannelZ | Does 'sip show peer xxx' show no Mailbox for that device? |
| 03:03.25 | drfreeze | Mailbox : |
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| 03:08.19 | ChannelZ | hmm |
| 03:09.18 | TJNII | How old is the firmware? |
| 03:09.45 | drfreeze | The phone is a Polycom 335. It had the letter in the screen and the scrolling message that 1 message was waiting |
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| 03:11.04 | drfreeze | TJNII: 3.2.1.0078 3.2.1.0054 3.1.3.0439 |
| 03:13.24 | TJNII | You could try the newer firmware. |
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| 03:13.44 | Jumpie | drfreeze you sure thats not a missed call? |
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| 03:15.24 | drfreeze | so, MWI lights up on a missed call? |
| 03:16.26 | TJNII | Seems like something easy to test... |
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| 03:17.24 | ChannelZ | so, just res_timing_pthread doesn't seem to work as a timing source for things like MeetMe anyway |
| 03:17.58 | ChannelZ | or maybe there's another piece I'm missing, but I get an 'unable to open psuedo device' anyway |
| 03:18.56 | drfreeze | TJNII: our phones are usually 'group' phones, so it is not uncommon to have hundreds of missed calls, but they never light up |
| 03:19.58 | bjhaid | whenever i make a call on my softphone, after i hang up and try to put another call through, my ubuntu machine refuses to make sounds, so i cannot transmit voice data, can anyone help please |
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| 03:25.24 | drfreeze | I'm thinking a ball ping hammer may turn off the mwi |
| 03:28.53 | [TK]D-Fender | drfreeze: remove your mailbox line from sip.cfg |
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| 03:31.21 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 03:32.46 | drfreeze | [TK]D-Fender: it's currently commented out. I removed the line and reloaded sip |
| 03:33.00 | drfreeze | sip show peer xxx still shows same results |
| 03:33.17 | drfreeze | But, I can't tell if the MWI is on or off. |
| 03:33.41 | [TK]D-Fender | drfreeze: reset the phone |
| 03:34.19 | drfreeze | ok, done |
| 03:34.36 | drfreeze | assuming reset means reboot |
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| 04:19.27 | KavanS | asterisk color console issue (v. 1.4.28) - I've modified /etc/init.d/asterisk to include TERM=linux as well I've exported the TERM=linux via the command line |
| 04:19.47 | KavanS | I still do not have cli color in asterisk shell - even though ls --color works great |
| 04:19.56 | KavanS | asterisk -c via command line shows color as well |
| 04:20.28 | Barometer | maybe ls is aliased to shut off the color |
| 04:20.44 | KavanS | Barometer, I'm talking about when I am in asterisk - via asterisk -rvvvvvv |
| 04:20.54 | KavanS | it is in white text - no color, no syntax highlighting |
| 04:21.30 | Barometer | you said ls --color works, so made me think that ls in asterisk was just the system ls |
| 04:22.01 | Barometer | which might borrow off the user shell profile |
| 04:22.08 | Barometer | or the system profile |
| 04:22.46 | *** join/#asterisk knot (yiffstar66@unaffiliated/devemo) |
| 04:22.48 | ChannelZ | Color won't work if you run asterisk detached |
| 04:23.00 | KavanS | uggh, it used to |
| 04:23.47 | ChannelZ | You can run safe_asterisk for instance though which 'wraps' the main asterisk process non-forking and will give you back color |
| 04:24.48 | *** part/#asterisk errr (~errr@fedora/errr) |
| 04:32.38 | KavanS | n/m found a l33t fix |
| 04:32.40 | KavanS | http://svn.dd-wrt.com:8000/dd-wrt/browser/src/router/asterisk/contrib/init.d/rc.debian.asterisk?rev=14109 |
| 04:32.57 | KavanS | couple modifications to init script and color is back - total win |
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| 04:46.18 | QN | hi. i have a problem. i have 4 sip trunk setup in the system, however, i can't seems to call the external number of the trunk that is configured within the same server. it'll just throw me with the standard "Busy Here" message. |
| 04:46.23 | QN | any ideas? |
| 04:46.29 | QN | thank you. |
| 04:47.10 | QN | sorry, what i meant was calling the sip number via an internal extension. |
| 04:47.21 | *** join/#asterisk QaDeS (~mklaus@p4FC71616.dip0.t-ipconnect.de) |
| 04:47.56 | p3nguin | Show me the dialplan. |
| 04:48.04 | p3nguin | Show me the sip debug. |
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| 04:59.18 | ruben23 | <PROTECTED> |
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| 05:03.24 | [TK]D-Fender | ruben23: There is no way using * |
| 05:04.42 | ruben23 | <PROTECTED> |
| 05:05.17 | [TK]D-Fender | rubThen go ask in a Skype channel |
| 05:06.38 | ruben23 | <PROTECTED> |
| 05:07.07 | [TK]D-Fender | ruben23: http://www.google.ca/search?hl=en&source=hp&q=unlimited+skype+out&aq=f&aqi=g1&aql=&oq=&gs_rfai= |
| 05:08.59 | ruben23 | <PROTECTED> |
| 05:12.54 | QN | p3nguin: http://pastebin.com/M3z4HCEt <-- the debug log 61234560 & 61234561 are the sip number involved. i used ext 983 to call 61234561, and the outgoing call route is via 61234560. as for the dialplan, i used 6. please assist. |
| 05:15.15 | p3nguin | Ah, now it's starting to make more sense -- FreePBX is involved. |
| 05:17.16 | [TK]D-Fender | ruben23: https://buy.skype.com/paymonthly/?country=US#|allSubscriptionsTab |
| 05:19.03 | [TK]D-Fender | QN: Contact: <sip:61234560@192.168.0.88> <-- you have not set your system up properly to work from behind NAT |
| 05:19.04 | [TK]D-Fender | ~sipnat |
| 05:19.05 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 05:19.07 | [TK]D-Fender | ^^^^^^^^^^ |
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| 05:24.09 | ruben23 | [TK]D-Fender:thanks i got it |
| 05:25.12 | [TK]D-Fender | ruben23: Amazing what you can find... what you actually look at THEIR SITE |
| 05:26.44 | QN | [TK]D-Fender: thanks for the advise, actually i replaced the original internet server ip address of the server to 192.168.0.88.. and i think i made it confusing.. sorry. |
| 05:27.06 | QN | http://pastebin.com/fXnqj3PN <-- is the original where i just replace it with words. |
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| 05:28.15 | p3nguin | Why are you replacing anything with anything other than what it is supposed to be? |
| 05:28.53 | QN | sorry for the confusion, i was initially confused on the output. |
| 05:28.56 | [TK]D-Fender | QN: No, that is pretty clearly NOT forged. You have not apid attention |
| 05:28.58 | [TK]D-Fender | paid* |
| 05:29.11 | [TK]D-Fender | QN: You are sending the wrong CONTACT to your ITSP |
| 05:30.07 | QN | uh.. however, if i called an external number it works. |
| 05:30.45 | [TK]D-Fender | times up, bed beckons... |
| 05:31.09 | *** join/#asterisk Professional (~Pro@unaffiliated/shani) |
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| 05:31.46 | QN | or have i wrongly configured the sip trunk's peer settings? |
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| 05:32.17 | p3nguin | ~freepbx |
| 05:32.18 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
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| 07:44.34 | jkroon | hi all, it seems some SNOM phones are sporadically having issues with Digest authentication - is it possible to re-activate basic auth for these specific extensions? The auth is on a (mostly) secure LAN and the particular client isn't too worried if the passwords are running plaintext on the wire. |
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| 08:02.46 | hajkym | exist some way how can i disable "call on hold" features? i comment in features.conf line: include => parkedcalls but not success |
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| 08:12.48 | ChannelZ | hajkym: not sure what you mean |
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| 08:17.22 | hajkym | ChannelZ: i want disable call on hold...i don't want to asterisk manage call on hold.. |
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| 08:18.33 | hajkym | when i press call on hold state in my voip sw asterisk try manage it... |
| 08:18.34 | WIMPy | Depends on your terminals, but usually they do it, not Asterisk. |
| 08:20.33 | ChannelZ | What is it you want to happen instead? |
| 08:21.29 | hajkym | nothing..i want to asterisk send "call on hold" sip message to callee UA |
| 08:22.12 | hajkym | i mean..SPD message "a=sendonly" |
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| 08:23.31 | hajkym | because when i send INVITE with SDP "a=sendonly" asterisk handle this message |
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| 09:17.50 | mallchin | hi, is it possible to over-ride the attended transfer feature on a per application basis? |
| 09:18.10 | mallchin | I'd like to turn it off for a particular context |
| 09:21.49 | dgd | please check http://www.webmin.com/ tell me if you are getting what looks to be a raw index |
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| 09:31.42 | Wimme | you can only transfer if you have t or T in your dial options, maybe for that context you can dial without using those? |
| 09:32.16 | Wimme | mallchin, |
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| 09:38.20 | mallchin | Wimme: unsure what you mean, sorry, new to transfers -- I don't think we use transfers at all, but for a particular contect I have an incoming call from another asterisk box, and they want to do transfers and have requested we disable it our end |
| 09:39.34 | mallchin | Wimme: I can see attended transfers are enabled in features.conf, just wondering how to ensure they're not used for a contect, would I need a t dial option defined in the context to handle them? and if it isn't defined they're not used for that context? |
| 09:40.23 | mallchin | Chainsaw: o/ |
| 09:42.57 | Chainsaw | Morning :) |
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| 10:08.03 | mallchin | morning :) |
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| 10:10.54 | DrDamnit | Running Asterisk SVN-trunk-r265227, and while in the directory, after I press the first three digits of the persons last name, asterisk crashes. Thoughts? |
| 10:18.34 | Wimme | on my mobile i can call a number, add a pause and dial another number, for example an extension, or a IVR choice, is it possible to do the same with dial() in asterisk? |
| 10:19.33 | Wimme | like dial(sip/to-trunk/*number*pause*exten*)? |
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| 10:22.08 | Wimme | do i need dial option D for it? |
| 10:25.12 | DrDamnit | Wimme: yes. Use a w |
| 10:25.18 | DrDamnit | need a long wait? use wwwwwwwwwwwwwwwwwwwwwwwwww |
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| 10:29.14 | Wimme | DrDamnit, how do i use the w? when i try D(2) it sends the 2 immediatly, in my ivr i start with with(1), so the 2 has been sent before the background()starts |
| 10:30.51 | DrDamnit | It the dial statement, you can make it wait by doing this: DIAL(ZAP/1/wwww1234567890) |
| 10:31.10 | Wimme | ye, just foud it |
| 10:31.12 | Wimme | tnx |
| 10:31.15 | DrDamnit | welcome. |
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| 10:51.39 | oktay | anybody familiar with the spa3102 ? |
| 10:52.02 | oktay | my experience with linux/unix is if somethign works.. it keeps working. |
| 10:52.14 | oktay | but with this asterisk,freepbx,spa3102 it's just the opposite |
| 10:52.22 | oktay | i get everything working.. two days later it stops working |
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| 10:53.26 | oktay | trying to figure out if it's the spa.. |
| 10:53.45 | QN | anyone could help me on a incoming fax problem? i could hear the fax tone when i dial into it, however it dosen't seems to receive fax from an external fax machine.. the line will just be picked up and after 30secs or so, it'll just disconnect. |
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| 10:59.06 | oktay | this channel is in USA tiem i think |
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| 10:59.20 | lantizia | Is asterisk aware of MAC addresses of peers? |
| 11:02.38 | kaldemar | lantizia: no |
| 11:03.06 | kaldemar | oktay: what stops working? |
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| 11:16.49 | doolittlework | If i send a call to a sip channel and is busy it will go to the next priority, but is there a way to setup a noop so i can see why it is busy? |
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| 11:23.01 | coppice | QN I guess you have not directed the call to the fax application |
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| 11:32.18 | doolittlework | quite today |
| 11:34.24 | doolittlework | join #asterisknow |
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| 11:40.29 | oktay | kaldemar: i configure the system. For instance I call an extension from a soft phone. It works. 2 hours later. It doesn't work anymore. Things like this. Not always the same things, but always something breaks. |
| 11:43.26 | kaldemar | you need to be more specific than "it". does asterisk die? does a call fail? how does it fail? what does asterisk's CLI say when a call fails if that's the case? what is the network setup like? etc. |
| 11:46.31 | DrDamnit | Is there anything wrong with this statement? exten => s,n,Dial(DAHDI/g63/${DB(fwd/${ARG1})}) |
| 11:46.59 | DrDamnit | I am getting this error: syntax error: syntax error, unexpected '/', expecting $end; Input: |
| 11:47.53 | kaldemar | DrDamnit: not by itself. what does ARG1 have in it? |
| 11:47.57 | DrDamnit | yes |
| 11:48.09 | DrDamnit | kaldemar: it used to work just fine when it was ZAP. |
| 11:48.54 | DrDamnit | kaldemar: dahdi channels configured to this: ; Span 2: WCT1/0 "Wildcard TE122 Card 0" (MASTER) B8ZS/ESF ClockSource |
| 11:48.54 | DrDamnit | group=0,12 |
| 11:48.54 | DrDamnit | context=inbound |
| 11:48.54 | DrDamnit | switchtype = national |
| 11:48.54 | DrDamnit | signalling = pri_cpe |
| 11:48.56 | DrDamnit | channel => 9-31 |
| 11:48.58 | DrDamnit | context = default |
| 11:48.59 | DrDamnit | group = 63 |
| 11:49.04 | DrDamnit | I am concered about the duplicate groups. |
| 11:49.06 | kaldemar | well, pastebin a CLI output of a call. |
| 11:49.09 | kaldemar | ~pb |
| 11:49.12 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 11:49.22 | DrDamnit | over 3 lines... got it. |
| 11:49.47 | kaldemar | DrDamnit: you know that group = 63 has nothing to do with channels 9-31, right? |
| 11:50.41 | kaldemar | DrDamnit: and neither does context = default. in chan_dahdi.conf (and zapata.conf) the parameters need to be above the channel lines. |
| 11:51.15 | DrDamnit | kaldemar: No. I didn't. I was under the impression (as it used to work with ZAP) that setting a group gave the drive the ability to "hunt" for an open channel. G hunted from 9-31, and g hunted from 31-9. |
| 11:52.10 | DrDamnit | kaldemar: i missed that. |
| 11:52.10 | DrDamnit | you're right. |
| 11:52.10 | DrDamnit | kaldemar: so, is it group 0 or group 12 for those channels? |
| 11:52.22 | kaldemar | DrDamnit: both, according to your config. |
| 11:52.41 | DrDamnit | ok, let me change it ONLY to g0, and retry.... |
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| 11:54.26 | oktay | kaldemar: I can't be very specific because the issues are different each time. But this instance as an example. I called a physical extension from a soft phone. It worked. 2 minutes later I tried again and it says "all circuits are busy now" |
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| 11:56.16 | DrDamnit | kaldemar: ok that fixed some errors, and threw some others. http://pastebin.com/D8rRzZag |
| 11:57.17 | oktay | PSTN Line Status: Hook State:Off |
| 11:57.29 | oktay | the SPA is probably staying Off the hook after the initial call. |
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| 12:00.55 | kaldemar | DrDamnit: your macros seem to be pretty much broken. did you write them yourself? and, you're still trying to dial DAHDI/g63/... (on line 32) which seems to be invalid. |
| 12:02.03 | DrDamnit | I got it fixed. |
| 12:03.05 | tzafrir_laptop | Set("DAHDI/9-1", "CHANTYPE= DAH") |
| 12:03.32 | DrDamnit | kaldemar: I saw the g63, which needed to be changed since I changed the dahdi-channels.conf. |
| 12:03.48 | tzafrir_laptop | DrDamnit, how do you set the CHANTYPE? First three characters of the dial string? |
| 12:04.20 | DrDamnit | I am using Dial(DAHDI/X/X) |
| 12:04.26 | tzafrir_laptop | g12 would give you that specific port. g0 - all "PSTN ports" (for some strange definition of PSTN) |
| 12:05.00 | tzafrir_laptop | Also: what exactly is span 1? |
| 12:05.15 | tzafrir_laptop | hmm... probably some 8 port analog card |
| 12:06.24 | DrDamnit | tzafrir_laptop: span 1 is a TDM800, span 2 is a TE122 |
| 12:06.46 | tzafrir_laptop | DrDamnit, how does that line ( Set("DAHDI/9-1", "CHANTYPE= DAH") ) look in the dialplan? |
| 12:07.06 | tzafrir_laptop | There's also a suspicious-looking space there |
| 12:07.36 | DrDamnit | let me find that for you... |
| 12:07.43 | DrDamnit | this dialplan is being upgraded from a zaptel plan... |
| 12:08.42 | DrDamnit | tzafrir_laptop: good eyes. There is a macro that parses the channel type to make a decision on where to send it (ZAP/IAX/SIP). I need to mod that. |
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| 12:10.07 | oktay | again. right now. everything works.. this is going to kill me :) |
| 12:11.53 | DrDamnit | tzafrir_laptop: I am assuming you are the same tzafrir who writes the DAHDI stuff. Thanks for all your work. |
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| 12:34.30 | oktay | i guess it's voltages and unreliable real-world analog things that are messing with me |
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| 12:50.03 | DrDamnit | oktay: what's your issue? |
| 12:50.17 | oktay | one minute worky, the next no :) |
| 12:50.37 | oktay | i think it's about detecting that the call has finished |
| 12:51.39 | oktay | it happens from soft-phone to soft-phone as well as soft to pstn |
| 12:51.40 | oktay | right now all of it works. |
| 12:51.40 | oktay | but that's the problem. I'd rather it never worked. |
| 12:51.40 | oktay | Than work one minute and stpo working later. |
| 12:52.41 | DrDamnit | what version of Asterisk are you using? |
| 12:52.54 | oktay | 1.6.2.0~rc2-0ubuntu1.2 |
| 12:52.58 | DrDamnit | that sounds like the native bridge is having an issue if it is SIP and Asterisk. |
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| 12:53.10 | DrDamnit | is this a production box? |
| 12:53.15 | oktay | yes |
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| 12:53.26 | oktay | there are many other things on it. mostly network related..f irewall/proxy etc. |
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| 12:54.13 | DrDamnit | you could try using the most recent svn. As a general rule, I tend to not put anything else on the Asterisk box. It just simplifies life. even if i compartimentalize it by putting it in a VM (only possible if you have no hardware). |
| 12:55.28 | oktay | VM is a possibility. We don't have hardware. |
| 12:55.43 | oktay | Is there a good packaged VM out there? |
| 12:55.45 | leifmadsen | 1.6.2.0-rc2 is SEVERAL months old |
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| 12:56.39 | oktay | DrDamnit: what do you mean by 'Native Bridge' ? |
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| 13:00.42 | QN | could anyone help me on a incoming fax problem? whenever there's a fax from external rings in, the line will just be picked up and after 30secs or so, it'll just disconnect. no transmission of fax at all. it seemes to me like the receiving fax server can't "hear" what has been put thru by the external fax machine.. |
| 13:01.10 | [TK]D-Fender | QN: Pastebin the failed call with verbose 10 |
| 13:01.15 | [TK]D-Fender | ~pb |
| 13:01.15 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
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| 13:09.16 | QN | [TK]D-Fender: http://pastebin.com/anpH74z8 |
| 13:10.27 | tuxx- | hiya guys, does asterisk support call waiting out of the box? What i mean by call waiting is the following: Customer A has a conversation with B. When C tries to call B, he hears a busytone, but wants to place the call behind B's conversation, so that when B hangs up, his phone will ring with the call that has just been placed by C. |
| 13:10.43 | oktay | your customers have funny names |
| 13:10.48 | tuxx- | :) |
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| 13:16.04 | oktay | cell phones sometimes work that way |
| 13:16.17 | oktay | i took a quick look but didn't see anything relating to that |
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| 13:22.02 | Baylink-away | tuxx-: That's called "camp on", and I'm sure you could implement it atop asterisk, but I don't know that any of the packaged PBXs do -- I think that FreePBX does not, frex, but I haven't actively looked for it. |
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| 13:22.34 | tuxx- | thanks :), gonna google for 'camp on' :) |
| 13:22.42 | ManxPower | Asterisk does not have a real "camp-on" feature, but you can emulate it pretty well. See "core show application retrydial" |
| 13:22.51 | oktay | take care guys |
| 13:23.00 | tuxx- | tyvm xD |
| 13:23.02 | Baylink-away | Thanx, Manx. |
| 13:23.12 | ManxPower | Baylink-away, Also try "core show applications" |
| 13:23.35 | Baylink-work | I wasn't the one who cared. :-) |
| 13:24.04 | ManxPower | Baylink-work, That command is useful to every single Asterisk user in the world. |
| 13:24.17 | Baylink-work | Sure. If you're running 1.4. :-) |
| 13:24.57 | ManxPower | that should work in 1.4 - 1.6.2+ |
| 13:25.26 | Baylink-work | Yeah; I got about 19 machines running 1.2.30.2, that will be for the foreseeable future. |
| 13:26.31 | ManxPower | on 1.2 it would me "show applications" |
| 13:26.57 | ManxPower | s/me/be |
| 13:27.06 | Baylink-work | Yes, I know. |
| 13:27.52 | DrDamnit | oktay: the "Native Bridge" is the function inside asterisk that joins channels. it lets SIP talk to IAX, IAX to DAHDI, etc... it sounds like that's where the problem is since your intermittent issue is across all resouce types. So, get the newest version of 1.6. |
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| 13:41.09 | [TK]D-Fender | QN: Nowhere in there is your system even trying to receive a fax, and FreePBX is NOT supported in here |
| 13:41.11 | [TK]D-Fender | ~freepbx |
| 13:41.35 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 13:42.41 | [TK]D-Fender | ^^^^ |
| 13:43.40 | *** join/#asterisk shader (~40846872@gateway/web/freenode/x-rkqdupdugrfbhsxk) |
| 13:44.00 | [TK]D-Fender | tuxx-: that depends. If you are referring to being on an analog line and that LINE receives a second call, then this is problematic. If you are referring to just being on the phone and getting another call then taht's up to the phone. In 99.99999% of cases the answer is "yes" |
| 13:44.01 | shader | does anyone here have experience configuring asterisk and dovecot to provide imap access to voicemail? |
| 13:44.27 | [TK]D-Fender | tuxx-: VoIP Phones do this by themselves and generally only shitty softphones may limit you on this. |
| 13:46.04 | tuxx- | [TK]D-Fender: sure, we have that feature on our aastra phone, its just not exactly what we're looking for. The application RetryDial helps alot :) |
| 13:46.09 | tuxx- | thanks anyway for the reply! :) |
| 13:47.19 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:48.52 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 13:49.52 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 13:50.24 | QN | [TK]D-Fender: i've piped hylafax at ext 984 and have set a incoming route to 984 directly. if i call from DID, i can hear fax tone. But no fax machine can ever actually connect. Any test fax reports failure. |
| 13:54.38 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 13:54.58 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 13:56.01 | [TK]D-Fender | QN: Have your extension ANSWER the call first before calling the device |
| 13:56.25 | *** join/#asterisk afink (~afink@204.26.87.226) |
| 13:56.55 | *** join/#asterisk c0rnoTa (~c0rnoTa@80.251.113.56) |
| 14:02.21 | [TK]D-Fender | QN: Also Fax over SIP like that is pretty much an assumed failure without T.38 |
| 14:04.33 | QN | [TK]D-Fender: DID -> Ring into Ext 984 -> Fax server at Ext 984 picked up and sounded fax tones -> Keep on looping the same thing for another 30-40 secs and the fax server assumes that there's no handshaking and hanged up. |
| 14:05.06 | [TK]D-Fender | QN: And as I said, Fax over VoIP can be assumed to fail without T.38 |
| 14:05.17 | Belgarath | DID is over VoIP od PTSN lines ? |
| 14:05.27 | QN | Belgarath: voip |
| 14:05.35 | Belgarath | no joy |
| 14:05.38 | [TK]D-Fender | Belgarath: It's been repeated twice... VoIP <- |
| 14:05.46 | Belgarath | go get fax2 mail gateway |
| 14:05.46 | *** join/#asterisk Slugs_ (~Slugs@unaffiliated/slugs) |
| 14:05.56 | Belgarath | [TK]D-Fender: sorry |
| 14:06.00 | Belgarath | just came in |
| 14:06.01 | Belgarath | :) |
| 14:06.13 | beek | mornin' [TK]D-Fender |
| 14:06.19 | QN | [TK]D-Fender: would like to know if there's way to have t.38 supported? |
| 14:06.34 | Belgarath | QN: yes but your provider has to support it first |
| 14:06.37 | Belgarath | go ask them |
| 14:06.38 | Belgarath | :P |
| 14:06.44 | [TK]D-Fender | QN: Go ask your provider, and go check how to do so in #freeppbx |
| 14:08.22 | QN | hmm.. t.38 is codec? |
| 14:09.01 | devmod | Has anyone successfully played aupix h323 video files ? http://www.aupix.com/index.htm?/support/astvideo.htm |
| 14:11.52 | [TK]D-Fender | QN: No, it is a protocol |
| 14:16.44 | ManxPower | Does anyone have recordings of "garbled" SIP calls? I need something to help our sales people understand the difference between "static" (impossible on VoIP must be a CPE issue) and "garbled" (packet loss/jitter) |
| 14:18.43 | Chainsaw | ManxPower: It degrades a lot like how a digital cellphone signal would. Have you made that comparison? |
| 14:19.21 | ManxPower | Chainsaw, Our sales people are just slightly smarter than a cockroach |
| 14:19.22 | kaldemar | ManxPower: make a call through IAX2, put some packet loss on it with 'iax2 test losspct' and record it. |
| 14:19.39 | ManxPower | So the answer is "no". |
| 14:20.51 | ManxPower | I can create such a recording, but wanted to know if anyone has one before I create a problem and record it |
| 14:21.31 | *** join/#asterisk eliel (~eliels@200.43.6.244) |
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| 14:39.46 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
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| 14:46.53 | *** join/#asterisk niekvlessert (~niekvless@92.70.112.34) |
| 14:47.10 | niekvlessert | hello! anybody knows about this in the full asterisk log? |
| 14:47.20 | niekvlessert | <PROTECTED> |
| 14:47.28 | niekvlessert | and then it goes off |
| 14:47.45 | niekvlessert | there are rumors this causes conversations to get lost |
| 14:49.10 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:49.10 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:52.36 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:53.25 | *** join/#asterisk emora (~emora@213.236.9.114) |
| 14:54.06 | emora | hello |
| 14:54.50 | emora | I have a question about a warning message thats really not affecting operation but I don't know why it shows up |
| 14:55.03 | emora | at reload we get a warning |
| 14:55.04 | emora | <PROTECTED> |
| 14:55.57 | emora | any suggestions on what I can do to make it go a away? |
| 14:56.19 | pabelanger | emora: Install your .config file? |
| 14:56.47 | emora | .config ? |
| 14:56.55 | *** join/#asterisk ldiamond (~chatzilla@24-226-250-193.tr.cgocable.ca) |
| 14:57.22 | pabelanger | emora: Failed to load configuration file <-- This is your problem. |
| 14:57.23 | emora | voicemail.conf is in place and working |
| 14:57.47 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:57.54 | file | app_minivm is not app_voicemail |
| 14:57.59 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 14:58.17 | *** join/#asterisk Tim_Toady (~moi@188.4.4.16.dsl.dyn.forthnet.gr) |
| 14:59.17 | emora | I guess if its not in use I would rather not load it at all. |
| 14:59.50 | [TK]D-Fender | emora: Go right ahead and noload > app_minivm.so in modules.conf |
| 15:00.25 | emora | thanks [TK]D-Fender |
| 15:00.25 | *** join/#asterisk amplisys (~amplisys@modemcable009.0-59-74.mc.videotron.ca) |
| 15:01.56 | *** join/#asterisk moy (~moy@bas1-unionville55-1177733627.dsl.bell.ca) |
| 15:03.17 | ldiamond | I am looking to setup a nice phone system for my home. I want to be able to take/place calls from my PC (need a nice, light-weight SIP application), use google voice, use my mobile phone through 3G to place calls using my SIP phone number and most of all, I want to save money! Now I am not sure where to start... seems like I need a SIP provider / SIP trunk? I am not very familiar with the... |
| 15:03.18 | ldiamond | ...terms, the costs, the good/bad companies, etc. |
| 15:03.38 | ldiamond | I want the calls to be clear and good quality, no delay, etc. |
| 15:04.58 | ldiamond | Please note that I live in canada, so I would like to have a canadian phone number that I can forward to Google Voice (or something similar). |
| 15:06.12 | [TK]D-Fender | ldiamond: go read up on gtalk w/ * on the WIKI and in the sample configs |
| 15:06.12 | [TK]D-Fender | ldiamond: as for what provider to use for termination.... shop around. |
| 15:06.26 | [TK]D-Fender | ldiamond: voip.ms , and les.net are 2 of the stronger choices for you |
| 15:06.47 | *** join/#asterisk BANSAL (~bansal@117.199.114.84) |
| 15:07.06 | [TK]D-Fender | ldiamond: And there is no point forwarding to GV... the calls would go direct to your * and from there you could call your SIP phones (and client on your cell) |
| 15:08.19 | ldiamond | Yes but GV would allow me to have down time on my * server |
| 15:08.35 | *** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br) |
| 15:09.05 | shader | I have ncurses installed, but menuselect isn't working |
| 15:10.26 | Infin1ty | i can put a call into a pickup to pick it up from another extension using AGI? |
| 15:11.12 | *** join/#asterisk retentiveboy (~pdugas@atl.pra-corp.com) |
| 15:11.27 | [TK]D-Fender | shader: ncurses-devel <--- |
| 15:11.52 | shader | [TK]D-Fender: I installed that too |
| 15:12.49 | [TK]D-Fender | shader: trash your extracted forder and do it again |
| 15:13.25 | shader | ok |
| 15:13.28 | [TK]D-Fender | Infin1ty: AGI = dialplan. It'll do whatever you tell it to.. |
| 15:13.33 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 15:13.48 | *** join/#asterisk Raden (~Raden@71.89.121.119) |
| 15:13.51 | *** join/#asterisk smooth_penguin (~smoove@115.118.146.198) |
| 15:13.55 | Infin1ty | oh, like Pickup() etc..? |
| 15:14.03 | Infin1ty | same as in extensions.conf? |
| 15:14.21 | *** join/#asterisk jshriver (~jshriver@cblmdm24-53-177-197.buckeyecom.net) |
| 15:14.25 | jshriver | Greetings. |
| 15:14.38 | [TK]D-Fender | Infin1ty: What are you expecting AGI to add to this process? |
| 15:14.48 | jshriver | How can I tell if the echo cancellation module is loaded and running in asterisk (software not hardware) having some horrible echo's in calls |
| 15:14.51 | Infin1ty | to manage a pickup extension |
| 15:14.56 | [TK]D-Fender | Infin1ty: AGI = call processing. What does it contribute that straight dialplan would not? |
| 15:15.09 | [TK]D-Fender | Infin1ty: How do you NORMALLY do a pickup? |
| 15:15.17 | Infin1ty | [TK]D-Fender, i don't know, that's why i asked, i didn't know :) in exten =>.... Pickup() etc.. |
| 15:15.27 | [TK]D-Fender | jshriver: "dahdi show channel [x] |
| 15:15.38 | Katty | looks in |
| 15:16.07 | [TK]D-Fender | Infin1ty: You don't put a call into "pickup". You can PARK a call, you can HIJACK one (AMI Redirect, etc), |
| 15:16.23 | Infin1ty | [TK]D-Fender, ye, i park a call then can pick it up |
| 15:16.30 | jshriver | Fender: that shows all of my channels but how can I tell if it has echo cancellation? |
| 15:16.40 | KavanS | nice |
| 15:16.42 | [TK]D-Fender | jshriver: [x] <----------- |
| 15:16.48 | KavanS | hijacking a call sounds real - how does one do this? |
| 15:16.52 | Katty | hello my asterisk does not work at all how to fix pls |
| 15:16.56 | KavanS | what search terms for teh google? |
| 15:17.00 | [TK]D-Fender | KavanS: Read what I told him |
| 15:17.09 | [TK]D-Fender | KavanS: I handed you both that answer |
| 15:17.10 | KavanS | understood |
| 15:17.14 | jshriver | ok none have an or even []. How do I enable dahdi echo cancellation |
| 15:17.15 | smooth_penguin | Katty, rinse, scrub, dry |
| 15:17.29 | Katty | shouldn't that be scrub, rinse, dry |
| 15:17.39 | Katty | else you might have soapy spots |
| 15:17.40 | smooth_penguin | what do men know |
| 15:17.41 | smooth_penguin | :P |
| 15:17.45 | beefpastry | Why does "host=dynamic" need to be there when trunking between boxes in iax? |
| 15:17.52 | jshriver | Echo Cancellation: 128 taps, currently OFF |
| 15:17.55 | beefpastry | just curious |
| 15:18.03 | Katty | what on earth is a beef pastry |
| 15:18.05 | jshriver | or if you can point me to some docs that would be appreciated. |
| 15:18.42 | Katty | oooh. beef stuffed pastry on google looks delicious |
| 15:19.41 | jshriver | or which echo cancellation module do you recommend? see there are several |
| 15:19.59 | [TK]D-Fender | beefpastry: Who said it had to be host=dynamic? |
| 15:20.35 | [TK]D-Fender | jshriver: And is that channel currently in use? |
| 15:20.39 | KavanS | chan_iax2.c:1178 __send_lagrq: I was supposed to send a LAGRQ with callno 1950, but no such call exists (and I cannot remove lagid, either). |
| 15:20.40 | jshriver | mg2, kb1, sec2, and sec |
| 15:20.44 | beefpastry | nobody, in theory, but I got: "Peer is not dynamic" in the log. |
| 15:20.47 | KavanS | any ideas on this thing dirtying up my console/logs ^^^ |
| 15:20.55 | jshriver | Fender I'll be bringing the asterisk server down to make the dahdi changes and restart it |
| 15:21.12 | beefpastry | switched it to dynamic and control it with deny/permit |
| 15:21.13 | jshriver | and if it works will schedule a maintainance time this evening for production server |
| 15:21.19 | ChannelZ | hey Katty wb |
| 15:21.20 | beefpastry | and registration works |
| 15:21.36 | *** join/#asterisk ManxPower (~manxpower@195.sub-75-216-192.myvzw.com) |
| 15:21.59 | [TK]D-Fender | beefpastry: Which would be beacuse you told one side to REGISTER and the other side said "This guy isn't <dynamic>, so WTF is he doing trying to REGISTER with us?" |
| 15:22.11 | [TK]D-Fender | beefpastry: So stop REGISTERING |
| 15:23.17 | Katty | hi ChannelZ (= |
| 15:23.24 | ManxPower | Registration notifies the remote side what ip is associated with that account. If you know your IP and it does not change, there is no point in registering. It has NOTHING to do with being able to place calls from a phone, only with receiving a call from Asterisk to your phone |
| 15:24.49 | Infin1ty | [TK]D-Fender, what i want to do is actually, dial in to an extension 1000 (for example) while 1000 is not registered park the call then wait for 1000 to register and when it registers so it will pick it up, is it possible with park and pickup? |
| 15:24.52 | jshriver | D-Fender: does the echocancellers=X,chan go in chan_dahdi.conf see the other dahdi conf is generated |
| 15:24.55 | ChannelZ | Where ya been? |
| 15:25.09 | beefpastry | So the problem is in defining the host along with a registration string. |
| 15:25.43 | ldiamond | Is there a SIP help channel? or maybe an explanation somewhere on how to setup a decent system? I am still trying to figure out what I need to buy... What is a trunk? is a DID all I need? or can I just buy a cheap number and use whatever provider I want with this number (sort of like a domain name) ? |
| 15:26.16 | *** join/#asterisk smooth_penguin (~smoove@115.118.146.8) |
| 15:26.29 | [TK]D-Fender | Infin1ty: What you would do is have the dailaplan for that exension check if the device is registered or not, and if not wait around a little and check agin. Repeat. When the device actually IS availab, jstu dial it. No Pickup, no nothing. |
| 15:27.04 | [TK]D-Fender | jshriver: chan_dahdi only uses "echocancel=yes. It doesn't define the type, on whether its used |
| 15:27.30 | [TK]D-Fender | ldiamond: How you get to the PSTN is up to you. How do YOU want to call out? |
| 15:27.55 | [TK]D-Fender | ldiamond: If you want to use an ITSP, then yes, a SIP provider would be it. |
| 15:27.57 | ldiamond | From my cellphone, over wifi, using SIPdroid |
| 15:27.58 | Infin1ty | [TK]D-Fender,will the caller still get a normal ring as this extensions is registered? |
| 15:28.15 | [TK]D-Fender | Infin1ty: Its your dialplan... do whatever you want. |
| 15:28.32 | jshriver | d-Fender where do you specify the echocancellers=mg2,1-8 then? the docs doesnt say |
| 15:28.40 | ManxPower | Infin1ty, I feel what you are trying to do is a bad idea. |
| 15:28.50 | [TK]D-Fender | jshriver: Yes, they do... system.conf <-- |
| 15:28.52 | Infin1ty | ManxPower, why? |
| 15:28.53 | ChannelZ | scampers off to work |
| 15:29.11 | [TK]D-Fender | Infin1ty: Ok, fine, sure |
| 15:29.17 | ManxPower | Infin1ty, Because you should not be hanging around waiting for phones to register. |
| 15:29.28 | jshriver | brb doc digging I dont even have a system.conf |
| 15:29.37 | ManxPower | jshriver, then you don't have DAHDI |
| 15:29.38 | Infin1ty | ManxPower, i do this for a purpose, i will hang it after X time |
| 15:29.39 | ldiamond | D-Fender, What do I need to get to the PSTN? |
| 15:29.39 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 15:30.02 | jshriver | yup I do , dahdi_scan etc.. works fine |
| 15:30.05 | ManxPower | jshriver, install DAHDI and try again |
| 15:30.08 | [TK]D-Fender | ldiamond: Depends HOW you indend on getting there |
| 15:30.17 | ManxPower | jshriver, you are looking in /etc/dahdi right? |
| 15:30.23 | [TK]D-Fender | jshriver: /etc/dahdi/system.conf <- |
| 15:30.25 | jshriver | nvm I see it, was looking in /mnt/kd/asterisk |
| 15:30.32 | ldiamond | [TK]D-Fender: well, I don't know lol |
| 15:31.05 | [TK]D-Fender | ldiamond: Then make up your mind. Did you have a physical line you wanted to use? |
| 15:31.09 | ManxPower | Infin1ty, I hope you are very good with Asterisk. What you want to do is something I've never heard of anyone (in their right mind) trying before. |
| 15:31.13 | jshriver | weird it shows per channel echocanceller=mg2 for each channel in the system.conf but when I do a dahdis show channel 8 for example says echo cancellation is off |
| 15:31.29 | ManxPower | jshriver, Do you have a call on channel 8? |
| 15:31.44 | jshriver | no, no calls coming/going right now |
| 15:31.45 | ManxPower | Why should Asterisk waste CPU echo canceling a channel that is not in use? |
| 15:31.48 | ldiamond | [TK]D-Fender: Well, it's not about making my mind, it's about me having no idea what the options are. No I don't have a physical line |
| 15:31.50 | Infin1ty | ManxPower, hmm, my main idea is to sms when an incoming call comes then wait for the device to register, just for testing right now |
| 15:31.53 | jshriver | oh |
| 15:31.54 | [TK]D-Fender | jshriver: Its only active while in a CALL |
| 15:31.59 | jshriver | ok let me give that a try, ty |
| 15:32.16 | ManxPower | Infin1ty, Why not keep phones registered? |
| 15:32.27 | Infin1ty | ManxPower, it's not possible, it's a mobile |
| 15:32.32 | [TK]D-Fender | ldiamond: If you have a cell phone, that can call the PSTN. If you have an analog line, then THAT can be used to call to the PSTN. If you have a PRI, then that, etc |
| 15:32.38 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 15:32.53 | ManxPower | Infin1ty, My mobile phone can always make and receive calls |
| 15:33.30 | ManxPower | Infin1ty, In any case, I think you are crazy, but I wish you the BEST of luck. |
| 15:33.35 | Infin1ty | ManxPower, because you are always registered to the extensions |
| 15:33.40 | ldiamond | [TK]D-Fender: But for example, if I use my cell phone to call the PSTN, I will be charged on my plan. I am trying to lower my phone bill. Therefore, I suppose I need a SIP provider ? |
| 15:33.56 | [TK]D-Fender | ldiamond: That is certainly an option. |
| 15:34.08 | *** join/#asterisk Polysics (~Luca@host88-205-dynamic.24-79-r.retail.telecomitalia.it) |
| 15:34.09 | [TK]D-Fender | ldiamond: I already passed you 2 providers to loko at |
| 15:34.10 | Polysics | hello |
| 15:34.11 | Infin1ty | ManxPower, for example, on an iphone it's not possible. |
| 15:34.25 | Polysics | i am experiencing some weird behavior on my * |
| 15:34.52 | Polysics | when someone hangs up on the caller without picking up first, the BUSY signal is sent |
| 15:35.04 | ldiamond | [TK]D-Fender: yea, I am looking into them right now, thanks. |
| 15:35.04 | Polysics | but the caller does not receive the BUSY until a few seconds after |
| 15:35.17 | *** join/#asterisk imox1234 (~imox1234@p4FC5C527.dip0.t-ipconnect.de) |
| 15:35.26 | ManxPower | Polysics, your question does not parse |
| 15:35.28 | [TK]D-Fender | Polysics: PASTEBIN |
| 15:35.40 | ManxPower | "someone" can't hangup a call unless "someone" answers a call. |
| 15:35.48 | Polysics | * complains about WARNING[9921]: channel.c:3186 ast_indicate_data: Unable to handle indication 5 for SIP/whatever |
| 15:35.59 | Polysics | ManxPower, they reject the call, bad term on my part |
| 15:35.59 | ManxPower | Polysics, you are missing /etc/asterisk/indications.conf |
| 15:36.18 | ManxPower | Polysics, They press the REJECT Soft key on their Polycom phone? |
| 15:36.37 | jshriver | thanks for the help, looks like it was already "working" |
| 15:36.42 | Polysics | they press the "red" button on X-LIte |
| 15:36.49 | Polysics | but i suppose it does the same thing |
| 15:36.59 | jshriver | is there a way to get echo cancellation to work more quickly? It sounds like it does eventually cancel it out but takes a good 15-30seconds before it kicks in |
| 15:37.00 | Polysics | x-lite or zoiper, same behavior |
| 15:37.01 | ManxPower | Polysics, fix your /etc/asterisk/indications.conf and try it again |
| 15:37.18 | jshriver | this is form a SIP phone -> asterisk -> pots local call |
| 15:37.38 | Polysics | so i suppose this has always happened but i never noticed? |
| 15:37.39 | ManxPower | jshriver, You are missing the interface info for the POTS |
| 15:37.46 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 15:37.55 | jshriver | sorry don't understand what that means |
| 15:38.05 | ManxPower | Polysics, let me know when you have fixed your /etc/asterisk/indications.conf and reload asterisk. |
| 15:38.33 | jshriver | oh TDM410 I believe on an asterisk card |
| 15:38.49 | Polysics | ManxPower, i actually have no indications.conf whatsoever - 1.6.11 if it helps |
| 15:38.50 | jshriver | TDM410P card with those little red FXO chips by digium |
| 15:38.53 | Polysics | i'll get a sample one |
| 15:38.55 | *** join/#asterisk michael-i (~michael-i@141.41.40.121) |
| 15:39.19 | ManxPower | jshriver, If you have hardware echo issues with a card contact Digium |
| 15:39.27 | jshriver | using software |
| 15:39.27 | ManxPower | Polysics, Missing is also "broken" |
| 15:39.35 | ManxPower | jshriver, Ah. sucks to be you. |
| 15:39.37 | jshriver | the dahdi mg2 |
| 15:39.43 | Polysics | i will start with the default one |
| 15:39.57 | ManxPower | Polysics, the default one should work just fine |
| 15:40.09 | jshriver | yes it does I can't stand working on phone systems no idea how it got dumped on me to maintain this poc |
| 15:40.15 | michael-i | Hi everyone. I'm porting AskoziaPBX to the powerpc architecture and have an issue with /dev/dahdi/ctl not being generated. The modules are loaded cleanly but it is not present. Any pointers? Kernel config to watch for? |
| 15:40.32 | ManxPower | michael-i, Dev questions are better asked on #asterisk-dev |
| 15:40.33 | michael-i | things show up properly in /proc ... but nothing in /dev |
| 15:40.52 | Polysics | ManxPower, got default indications.conf off the wiki and restarted * |
| 15:41.06 | michael-i | ManxPower: I usually take a stab here first to be safe. Will do |
| 15:41.25 | ManxPower | Polysics, NO! NO!NO! |
| 15:41.35 | Polysics | eeks :-) |
| 15:41.49 | ManxPower | Polysics, Get the default indications from the place where the default indications file lives /path/to/src/asterisk/configs |
| 15:42.10 | ManxPower | Polysics, the wiki is the WORST PLACE to get config files |
| 15:42.37 | ManxPower | cp /path/to/src/asterisk/configs/indications.conf.sample /etc/asterisk/indications.conf |
| 15:43.06 | Polysics | done, restarting * |
| 15:43.36 | Polysics | ok, up now |
| 15:44.06 | Polysics | do i just try again? |
| 15:44.08 | ManxPower | See if you continue to get that error |
| 15:44.19 | *** part/#asterisk jshriver (~jshriver@cblmdm24-53-177-197.buckeyecom.net) |
| 15:47.09 | ManxPower | Polysics, at this point we should have eliminated the indication error. I suspect that will fix your other issue too. |
| 15:47.47 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 15:48.12 | Polysics | ManxPower, at the moment i hear a busy tone but the caller has to hang up manually - is that the intended way to work? |
| 15:48.31 | Polysics | no errors |
| 15:48.37 | [TK]D-Fender | Polysics: then you'd better copy over the sample for it |
| 15:48.50 | [TK]D-Fender | Polysics: Then you'd better copy over the sample config |
| 15:49.08 | shader | asterisk isn't connecting to my imap server because it's using a self-signed certificate. Is there a way to bypass that check? or get it to accept the cert? |
| 15:49.11 | [TK]D-Fender | Polysics: PASTBIN <- |
| 15:49.54 | Polysics | [TK]D-Fender, i have nothing to pastebin at the moment, as this is probably how asterisk is SUPPOSED to work ,it's me that needs a different behavior :-) |
| 15:50.48 | Polysics | i probably need to react to the Dial event that the BUSY triggers by killing the call |
| 15:52.22 | Polysics | sounds like a plan, unless tehre is a way to force hangup on BUSY in * |
| 15:53.15 | ManxPower | Polysics, you have to handle all response |
| 15:53.35 | ManxPower | If you ever wonder why Verizon sucks, here is an example: ...tech from VZ called stating that he is at the clients site to do the work but can't locate the DMARC and is going to leave.... |
| 15:54.07 | ManxPower | Polysics, you will need to do this with all non-FXO ports |
| 15:55.10 | Polysics | ManxPower, i think i am lost here |
| 15:55.21 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 15:55.26 | ManxPower | Polysics, look at macro-stdexten on extensions.conf.sample. |
| 15:55.28 | ldiamond | Is there any free outbound sip provider? I just want to be able to test my setup. (even if its max 1min calls or something like that) |
| 15:55.53 | ManxPower | ldiamond, why don't you spend $10 and get a PAYG account with some provider? |
| 15:56.26 | [TK]D-Fender | ldiamond: just call a SIP phone on your system or something then |
| 15:56.30 | ldiamond | ManxPower: I just want to be able to test my account before choosing the right provider |
| 15:56.54 | ManxPower | ldiamond, if you can't afford $10 then you should not be doing telecom |
| 15:56.58 | [TK]D-Fender | ldiamond: "TEST" is by and large a meaningless concept unless youa re using the precise solution you will in the end |
| 15:57.33 | [TK]D-Fender | ldiamond: The 2 I agve you are pretty well respected and many businesses coming through here are using them quite happily |
| 15:59.08 | *** join/#asterisk Janos (~cramos@190.10.52.113) |
| 15:59.09 | *** join/#asterisk Yon (Yon@2002:50d9:f410::50d9:f410) |
| 15:59.26 | ManxPower | ldiamond, http://ebooks-it.org/e-books/wiley/Wiley.VoIP.For.Dummies.Sep.2005.ISBN.0764588435.pdf |
| 16:00.56 | ldiamond | thx, ill read that |
| 16:01.17 | ManxPower | ldiamond, something is wrong with that link. Sorry |
| 16:01.30 | ldiamond | yea well, I got it somewhere else |
| 16:01.33 | ManxPower | ldiamond, let me check something first |
| 16:02.06 | ManxPower | I have the avaya version of that in dead tree format somewhere] |
| 16:03.41 | *** part/#asterisk qxork (~qxork@adsl-70-54-28.gnv.bellsouth.net) |
| 16:03.51 | ManxPower | [TK]D-Fender, and everyone else, ~voipfd is the infobot factoid for the (working) link to the VoIP for Dummies e-book |
| 16:05.55 | *** join/#asterisk Yonn (Yon@c80-217-244-16.bredband.comhem.se) |
| 16:06.29 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:07.01 | *** join/#asterisk timeshell (~timeshell@gw.lusi.on.ca) |
| 16:08.20 | *** join/#asterisk p3nguin (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 16:09.24 | *** join/#asterisk smooth_penguin (~smoove@219.64.166.125) |
| 16:09.25 | *** join/#asterisk kartik (~koolkarti@117.199.114.111) |
| 16:14.05 | *** join/#asterisk idespinner (~idespinne@cpe-76-93-115-243.socal.res.rr.com) |
| 16:14.56 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 16:17.19 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 16:17.25 | bmoraca_work | god i'm glad floppy disks aren't used anymore |
| 16:20.13 | DrDamnit | anyone ever use www.phoneprovisioning.com to setup phones? |
| 16:21.18 | thehar | i parsed that as floppy dicks |
| 16:22.21 | Chainsaw | bmoraca_work: Hard-sectored 5.25" HD wasn't so bad. |
| 16:22.25 | leifmadsen | DrDamnit: never used it, but think it's a neat idea... just not sure I like the idea of having personal information there so probably wouldn't use it for a large number of phones, but for home phones it might be useful |
| 16:22.45 | Chainsaw | bmoraca_work: 3.5" though, a few quick formats in a row and there was trouble. A lot of trouble. |
| 16:22.51 | leifmadsen | and now, I lunch |
| 16:23.07 | Chainsaw | Have a good one leif. |
| 16:23.36 | DrDamnit | seems secure. you can define that only your network IPs can access the information. |
| 16:24.14 | leifmadsen | until someone compromises their systems :) |
| 16:24.30 | paulc | true that |
| 16:25.06 | [TK]D-Fender | DrDamnit: 3rd party in possession of auth to your servers? YUCK |
| 16:25.06 | *** join/#asterisk sulex (~sulex@firewall.blindata.ch) |
| 16:25.19 | DrDamnit | so...an encrypted database would be better, yeah? |
| 16:25.43 | [TK]D-Fender | DrDamnit: DIY <- |
| 16:25.58 | leifmadsen | in fact, compromising that server (if it was popular even a little bit) could be a gold mine to authentication information for free phone calls |
| 16:26.16 | DrDamnit | www.phoneprovisioning.com is mine. Looking for ideas to improve it. |
| 16:26.35 | paulc | reminds me of "Voxilla At Your Service".. hmm.. www.voxilla-ays.com maybe? |
| 16:26.43 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 16:26.43 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:26.51 | ManxPower | I normally set my SIP secrets to be [mush the keyboard until there are enough characters] |
| 16:26.55 | paulc | they were doing it more for service providers.. you're "ITP Blah" and you send requests to them, they generate a config profile then ship the device to the end user |
| 16:27.05 | leifmadsen | I mean, I really like the idea, but I'm not sure I could get around the security implications. Now if it was easy to grab what you're doing and put it on an appliance or something on my local network, that might be very useful |
| 16:27.23 | DrDamnit | I have an appliance version. |
| 16:27.33 | DrDamnit | runs on linux boxes with the alix chipset. |
| 16:27.40 | leifmadsen | that seems neato |
| 16:27.54 | leifmadsen | phone provisioning always seems to be a problem for some people |
| 16:27.56 | DrDamnit | So, I'll put a link to where you can get the appliance version. |
| 16:28.23 | DrDamnit | It really is. The people from Polycom like the site. (On of the product managers contacted me), so I am looking to improve it. |
| 16:28.29 | Corydon76-dig | ManxPower: I generally prefer secrets to be a bit more deterministic; usually I take the username, add a salt, and MD5 hash it to get a password |
| 16:28.31 | DrDamnit | and adding Snom and Cisco support. |
| 16:29.15 | Corydon76-dig | 128 bits of what appears to be random is pretty good for password security |
| 16:30.17 | Corydon76-dig | and then I use the same salt for an entire site, making the passwords retrievable, even if there was a catastrophic failure at the site |
| 16:32.03 | Naikrovek | i have some perl scripts i need to toss up on a pastebin or something |
| 16:32.07 | Naikrovek | makes polycom provisioning easy |
| 16:32.15 | Corydon76-dig | The salt is usually some random phrase. I used "The Lord's Prayer" at a Christian site once. |
| 16:32.17 | Naikrovek | though certainly not perfect for everyone |
| 16:33.02 | leifmadsen | Naikrovek: not all condoms fit the same... wait, maybe that's not an appropriate analogy |
| 16:33.21 | ManxPower | Heh! Verizon came to install a line at a customer location and someone had cut the fiber at DMARC and stole the fiber termination equipment |
| 16:33.53 | Naikrovek | leifmadsen: heh |
| 16:34.14 | Corydon76-dig | leifmadsen: plus there's latex allergies... |
| 16:34.19 | [TK]D-Fender | ... |
| 16:34.24 | ManxPower | I always have to get the extra-large ones. |
| 16:34.27 | Naikrovek | well my system works for me. it works for all the polycom phones i have, and it's one script and one file. run the script, it reads the file, barfs out piles of phone configs |
| 16:34.34 | Naikrovek | couldn't be much easier for me |
| 16:34.42 | leifmadsen | Naikrovek: ya I have the same type of approach at a couple of sites |
| 16:34.51 | leifmadsen | just a file with a list of mac addresses |
| 16:35.14 | Naikrovek | mac, extension, name, linekeys, custom ringer file, sip passwd |
| 16:36.20 | Naikrovek | one line per phone. |
| 16:36.28 | Naikrovek | i suppose it's about as simple as possible |
| 16:36.39 | Naikrovek | i want to get fancy |
| 16:36.54 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 16:36.54 | Naikrovek | set up a webapp of some kind where configs are generated dynamically |
| 16:37.02 | Naikrovek | so i can skip the "regen the config files" step |
| 16:37.05 | Naikrovek | just edit a file, reboot the phone |
| 16:38.45 | devmod | what does video flag refers to in : "** Our capability: 0x80004 (ulaw|h263) Video flag: False Text flag: True" |
| 16:39.08 | *** join/#asterisk rocksfrow_work (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 16:39.15 | leifmadsen | videosupport=yes not enabled in sip.conf perhaps? |
| 16:39.20 | leifmadsen | throws out a wild guess |
| 16:40.37 | drmessano | I just love an app supports MSSQL only.. Hopefully Asterisk won't hate me too much for it |
| 16:40.52 | devmod | leifmadsen: its enabled |
| 16:40.52 | *** join/#asterisk nateflex (~ndavidson@c-174-52-20-94.hsd1.ut.comcast.net) |
| 16:41.02 | drmessano | pets Asterisk and give it an extra treat |
| 16:41.26 | rocksfrow_work | hears asterisk purring |
| 16:42.20 | drmessano | I need a cloud. I hear they're all the rage |
| 16:42.22 | Corydon76-dig | actually thinks Asterisk does binkies in the middle of the floor |
| 16:42.27 | *** join/#asterisk jart (~jart@c-76-23-206-246.hsd1.ct.comcast.net) |
| 16:42.40 | rocksfrow_work | drmessano: i'm building one :-) |
| 16:43.04 | Corydon76-dig | aims a humidifier at drmessano |
| 16:43.10 | drmessano | I have a bunch of Weather data I need to archive, then access with Asterisk. I wonder if I can get a cloud from eBay so i won't need to use any real hardware... just the Unicorn kind |
| 16:43.56 | drmessano | sed 's/Fancy Server/Invisible Cloud/' |
| 16:44.18 | leifmadsen | Lou Diamond Philips?! |
| 16:44.19 | Corydon76-dig | drmessano: I'll sell you an Imaginary Cloud |
| 16:44.29 | drmessano | Corydon76-dig: SOLD |
| 16:44.39 | leifmadsen | I'll sell you a Real Imaginary Cloud(tm) |
| 16:45.15 | Corydon76-dig | I'll sell you a Real Imaginary Cloud with a CDROM drive! |
| 16:46.26 | drmessano | Now I have the song "I need a Cloud with a well documented API" stuck in my head. Not that the song actually exists. |
| 16:47.02 | drmessano | Though if the Cloud is invisible and imaginary, the API should be easy.. API() |
| 16:47.42 | Corydon76-dig | drmessano: cf The Arrogant Worms "Wouldn't It Be Great If Everybody Had A Gun?" |
| 16:48.19 | *** join/#asterisk jonny330 (~IceChat7@12.222.63.34) |
| 16:48.44 | *** join/#asterisk emora (~emora@213.236.9.114) |
| 16:48.58 | drmessano | HA.. nice |
| 16:50.14 | *** join/#asterisk sat-man (~jlupresto@c-174-52-20-94.hsd1.ut.comcast.net) |
| 16:50.35 | sat-man | slaps nateflex around a bit with a large trout |
| 16:50.51 | Corydon76-dig | Oh, my mistake, the song is actually named "Let There Be Guns" |
| 16:51.46 | drmessano | That song would be perfect for the MoH on my "Delivery Guy Location Service" |
| 16:51.52 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 16:52.00 | drmessano | Scare the crap out of them before they even find the house |
| 16:52.24 | leifmadsen | :) |
| 16:53.07 | Corydon76-dig | Pick up one of their albums... any of them... |
| 16:54.37 | drmessano | The Coffee Song is hilarious |
| 16:55.35 | Infin1ty | i'm still trying to find the best way to do this, when someone call to an extension or a call is coming from a specific sip trunk, i send some notification to a user via an external app, then i want the caller that calls will have some kind of playpack for around 30 secs or until the specific extension registers, what's my best way? |
| 16:55.41 | p3nguin | Do you locate the delivery guys for customers, or do you help the delivery guys find the place they are going? |
| 16:56.53 | *** join/#asterisk diegomad (~mad@190.146.200.120) |
| 16:56.57 | Kobaz | do de do |
| 16:57.05 | Kobaz | anyone familiar with setting up lldp for polycoms |
| 16:57.14 | drmessano | Delivery guy calls and can't find my place. Send him to Allison announcement that the Outside Lights will be flashed to assist in locating the place, hangup when done. Then I flash my porch light with X10 |
| 16:57.34 | Infin1ty | Answer, background(<playback>) , wait 30 secs or until <extension> registers, is it possible? |
| 16:57.38 | drmessano | MoH during the alternating 3 second on/off states |
| 16:58.49 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 16:58.54 | [TK]D-Fender | Infin1ty: What I told you before + call out and send a Jabber alert or something. |
| 16:59.00 | leifmadsen | Infin1ty: I'm thinking you could use the MusicOnHold() application with a duration of like 15-20 seconds, then when it exists, have it check the peers registration status with SIPPEER() function, and if not registered, Playback() a message saying, "please wait while we find that extension" or somehting, then go back to MusicOnHold() for 10 more seconds, and loop for a bit. You could even limit the loop to a few times thro |
| 16:59.00 | leifmadsen | ugh (60 seconds) and then fail gracefully |
| 16:59.04 | [TK]D-Fender | infnet = +1 line fo dialplan to send the message |
| 16:59.14 | drmessano | Only PITA is I can't get state info from the X10, so I had compromise. h extension turns the light on, on the odds that I wouldnt be flashing a porch light in the daytime and probably want it ON when done |
| 16:59.36 | leifmadsen | and ya, if you need a notification, do what [TK]D-Fender just suggested with JabberSend() |
| 16:59.38 | *** part/#asterisk rocksfrow_work (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 16:59.50 | Kobaz | jabbar |
| 16:59.54 | Kobaz | arrr |
| 16:59.57 | drmessano | JabberSend rocks |
| 17:00.08 | leifmadsen | Pizza the Hut |
| 17:00.21 | Kobaz | leifmadsen: hah |
| 17:00.22 | leifmadsen | if you can be clever, you can dialplan |
| 17:00.46 | [TK]D-Fender | leifmadsen: RASPBERRY! Only 1 man in the Universe would DARE give me RASBERRY! |
| 17:01.21 | leifmadsen | so there you go: JabberSend(), MusicOnHold(), Playback(), and SIPPEER() should be all you need to accomplish what you want |
| 17:01.25 | drmessano | My calls hit Asterisk via Gizmo5, send the notification with JabberSend, and 5 seconds later my AT&T cell rings |
| 17:01.27 | *** join/#asterisk yang (yang@freenode/sponsor/cacert.assurer.yang) |
| 17:01.29 | leifmadsen | and NOW I go to lunch finally |
| 17:01.58 | Infin1ty | [TK]D-Fender, is there a good way to use some kind of wait until? like it has a background playback (i was able already to send a notification :) ) until the extension register? |
| 17:02.00 | drmessano | Run an Jabber client on the BlackBerry and I always have a nice heads-up |
| 17:02.24 | drmessano | You could park the call |
| 17:02.30 | drmessano | Send the notification with the parking slot |
| 17:02.54 | *** join/#asterisk Sipster_ (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 17:02.54 | bmoraca_work | WOW my network is slow today |
| 17:03.11 | drmessano | bmoraca_work: Unplug the other hub |
| 17:03.29 | bmoraca_work | i wouldn't be surprised if one of the other techs plugged a hub in |
| 17:03.39 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 17:04.00 | bmoraca_work | smarts is not their game when it comes to networking, unfortunately |
| 17:04.02 | drmessano | I miss hubs.. When bandwidth was all theory |
| 17:04.29 | bmoraca_work | i'm getting 600kbps to a server down stairs |
| 17:04.30 | bmoraca_work | wtf |
| 17:04.34 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 17:04.35 | [TK]D-Fender | Infin1ty: Dial a local channel, let that do teh waiting and dialing. In your Outer Dial() use m() to set a music class with your message |
| 17:05.32 | bmoraca_work | there we go, now it's picking up a bit |
| 17:05.35 | Infin1ty | [TK]D-Fender, erm, i still didn't understand, i will try both ways to find out by reading right now, it's just rtfm i guess |
| 17:05.37 | p3nguin | He'll never get it -- he thinks extensions register. |
| 17:05.56 | Infin1ty | a device register |
| 17:06.01 | [TK]D-Fender | p3nguin: Don't go all Grammar-Nazi on this.... |
| 17:06.11 | p3nguin | You lost me. |
| 17:06.15 | Infin1ty | p3nguin, it was just a shortcut for example of what i tried to achieve |
| 17:06.18 | [TK]D-Fender | Long ago |
| 17:08.37 | *** join/#asterisk timeshell (~timeshell@gw.lusi.on.ca) |
| 17:09.37 | bmoraca_work | we could start calling them ephones (the device) and ephone-dns (the equivalent of asterisk's peers) |
| 17:09.50 | bmoraca_work | yay for cisco! |
| 17:09.58 | drmessano | slaps bmoraca_work with a UC520 |
| 17:10.41 | bmoraca_work | point is: every modern telephony platform treats them differently, and legacy telco all refered to the phone as well as the number dialed to reach that phone as an extension |
| 17:11.05 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 17:11.56 | p3nguin | I believe it has been a misnomer all along. |
| 17:12.24 | p3nguin | Perhaps it takes an analytical person to understand. |
| 17:12.27 | bmoraca_work | that's not up to you and not subject to your beliefs. the industry has been around a lot longer than you have. |
| 17:12.57 | bmoraca_work | p3nguin: asterisk is the only telephony platform that calls dialplan entry points "extensions". no one else does that. |
| 17:13.15 | drmessano | I think users.conf and extensions.conf are misnomers if you're referring to usage |
| 17:13.29 | bmoraca_work | so, in the context of Asterisk, it makes sense to distinquish. but ONLY in the context of asterisk (and it's derivatives) |
| 17:17.12 | devmod | is videosupport=always still a valid option on 1.6.2 ? It breaks my video support whenever I set it. On the other hand using videosupport=yes works |
| 17:19.03 | *** join/#asterisk dajhorn (~dajhorn@adsl-75-17-124-26.dsl.rcsntx.sbcglobal.net) |
| 17:24.02 | leifmadsen | was videosupport=always ever an option? |
| 17:24.08 | leifmadsen | I always thought it was yes or no |
| 17:26.42 | DrDamnit | <PROTECTED> |
| 17:28.20 | leifmadsen | DrDamnit: that was quick! nice! |
| 17:29.10 | leifmadsen | DrDamnit: your online service supports provisioning via FTPS and/or HTTPS? |
| 17:29.16 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 17:29.24 | leifmadsen | apologies if that is mentioned on the site, I haven't had time to read through it all |
| 17:29.55 | leifmadsen | DrDamnit: I think you're missing the <H1> </H1> wrappers for your first title on the page too |
| 17:30.08 | leifmadsen | (or whatever you use for titles) |
| 17:30.31 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 17:36.04 | leifmadsen | DrDamnit: I suggest you re-think your yellow on blue schema :) |
| 17:36.43 | leifmadsen | DrDamnit: maybe the darker blue from the title "phoneprovisioning.com" with white text |
| 17:37.10 | DrDamnit | all good suggestions. Will do. |
| 17:37.27 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 17:37.30 | idespinner | DrDamnit, how are you going to keep the site up? |
| 17:37.34 | p3nguin | While we're on critiquing the page, "let's" should not have an apostrophe in it. |
| 17:37.44 | idespinner | i mean, cost wise. |
| 17:38.28 | DrDamnit | leifmadsen: currently, it only supports unencrypted http (it relies on IP filtering for security) because when I did the developing https didn't work very reliably on the Polycoms. Maybe because it requires a full ssl cert, and I was just using self-signed. |
| 17:38.41 | leifmadsen | gotcha |
| 17:38.49 | DrDamnit | idespinner: that site lives in a data center, on a RAID 5 box, with daily backups. |
| 17:38.54 | leifmadsen | probably worth it to support some sort of encrypted data though |
| 17:39.12 | DrDamnit | probably. There are several hundred users on there, but I have only had the request once or twice. |
| 17:39.35 | DrDamnit | biggest request I had was originally it didn't support configurable timezones (very annoying) I fixed that in a hurry. :-) |
| 17:39.51 | idespinner | DrDamnit, i'm just looking for how the site 'stays in business'. Surely you wont pay data center costs out of pocket forever :) |
| 17:40.28 | [TK]D-Fender | [13:38]<DrDamnit>leifmadsen: currently, it only supports unencrypted http (it relies on IP filtering for security) because when I did the developing https didn't work very reliably on the Polycoms. Maybe because it requires a full ssl cert, and I was just using self-signed. <- Huge trust issue here |
| 17:40.29 | DrDamnit | right now I do. It doesn't cost me much since it is on a box with other websites that generate recurring monthly revenue. |
| 17:40.55 | DrDamnit | TKD-Fender: thus... the appliance is now available. :-) |
| 17:41.12 | *** join/#asterisk nitram (foo@superblob.com) |
| 17:41.22 | [TK]D-Fender | DrDamnit: Hope you find a market. There ar e plenty of free scripts and FreePBX addons, etc for this already |
| 17:41.42 | DrDamnit | the site is free. |
| 17:41.43 | *** part/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 17:42.06 | DrDamnit | donations accepted, and really it is not a "business" so to speak. Just my donation to the world at this point. |
| 17:42.12 | [TK]D-Fender | DrDamnit: As is your "appliance"? |
| 17:42.48 | [TK]D-Fender | DrDamnit: Or are you subverting the term to imply you are distributing a full ISO to install this as a dedicated provisioning server? |
| 17:42.51 | DrDamnit | [TD]D-Fender: the appliance was the original product. I took the code from the appliance, and put it on the website as a free version because there is not a really good endpoint manaer for Polycoms. |
| 17:43.13 | DrDamnit | No. You get a plug and play box in the mail. |
| 17:43.16 | DrDamnit | or fedex. |
| 17:43.20 | DrDamnit | no ISO. |
| 17:43.35 | [TK]D-Fender | DrDamnit: You'll send me a plug and play box for FREE? |
| 17:43.46 | DrDamnit | of course not. The site is free, the appliance costs $$ |
| 17:43.50 | DrDamnit | :-) |
| 17:43.52 | DrDamnit | good try though. |
| 17:44.15 | paulc | How many $$? |
| 17:44.20 | leifmadsen | it's on the site |
| 17:44.21 | [TK]D-Fender | DrDamnit: So I have a choice of an unsecure free web-based service (increasing exposure), or a charged extra box to maintain locally? |
| 17:44.27 | DrDamnit | yes. |
| 17:44.39 | [TK]D-Fender | DrDamnit: I'll take door # 3 thank you..... |
| 17:44.43 | DrDamnit | hahahahaha |
| 17:44.46 | DrDamnit | everyone here would. |
| 17:45.00 | leifmadsen | the other door is setting it up yourself, which you're welcome to do |
| 17:45.08 | paulc | $247 + shipping, was the answer I was looking for |
| 17:45.29 | drmessano | new secuirty feature <---- |
| 17:45.33 | DrDamnit | The box itself is rather expensive in my opinion, but as part of a larger system it's a drop in the bucket, and end users love it. |
| 17:45.39 | drmessano | At least it's secuire |
| 17:47.24 | [TK]D-Fender | [13:44]<leifmadsen>the other door is setting it up yourself, which you're welcome to do <-- Happily doing so 5 years running :) |
| 17:47.43 | [TK]D-Fender | Dr(tm) |
| 17:47.47 | [TK]D-Fender | drmessano: (tm) |
| 17:47.54 | drmessano | Sorry |
| 17:47.56 | leifmadsen | [TK]D-Fender: yep, but not everyone necessary has the skills, knowledge, or time to do it, so I can see this would be advantageous to some people |
| 17:47.58 | drmessano | Secuire(tm) |
| 17:49.02 | DrDamnit | leifmadsen: it's basically like a linux router. Sure you can setup iptables no problem, but most end users would rather point and click, and hit save. |
| 17:49.13 | DrDamnit | s/linux/linksys |
| 17:49.35 | p3nguin | so close. |
| 17:50.10 | *** join/#asterisk roe (~roe___@unaffiliated/roe) |
| 17:51.52 | ManxPower | I wish outlook could do a rule: if subject = "" then reply "use a subject you idiot" |
| 17:51.59 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 17:52.32 | p3nguin | Now it just does "re: " doesn't it? |
| 17:52.34 | devmod | leifmadsen: videosupport [yes|NO|always] http://www.asterisk.org/doxygen/trunk/Config_sip.html |
| 17:52.52 | ManxPower | p3nguin, it won't match for empty subject as far as I can tell |
| 17:52.53 | leifmadsen | I'd check the code then to see if it is still supported |
| 17:53.11 | ManxPower | devmod, Are you running Trunk SVN? |
| 17:53.36 | devmod | ManxPower 1.6.2.6 actually |
| 17:54.17 | devmod | leifmadsen: yeah I see the code for it but it doesnt seem to work |
| 17:55.16 | *** join/#asterisk defslap (~andy@defsdoor.gotadsl.co.uk) |
| 17:56.46 | *** join/#asterisk fabrini (~fabrini@173-164-170-235-SFBA.hfc.comcastbusiness.net) |
| 17:59.17 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 17:59.44 | ManxPower | devmod, So you are looking at docs that are NOT for your version of Asteris |
| 18:01.48 | *** join/#asterisk lirakis (~etamme@65.200.191.241) |
| 18:04.35 | devmod | ManxPower: that information also comes on the sip.conf.sample included on the tarball I downloaded |
| 18:05.24 | bmoraca_work | DrDamnit: you can get SSL certs for ~$20/yr now |
| 18:07.00 | lirakis | bmoraca_work, fro go daddy? |
| 18:07.01 | *** join/#asterisk Arsenick (~y@modemcable230.231-70-69.static.videotron.ca) |
| 18:07.04 | lirakis | *from |
| 18:07.24 | bmoraca_work | company called SBS, i believe. I buy them through BulkRegister |
| 18:07.32 | [TK]D-Fender | devmod: So you set it to "yes" and it "doesn't work"? |
| 18:08.10 | devmod | [TK]D-Fender: If I set it to yes it works. If i set it to always it does not. |
| 18:09.31 | [TK]D-Fender | devmod: Well yes means yes... I'm not sure what to assume about "always". Whats the differnce between "yes" and "always" supposed to be? |
| 18:09.48 | [TK]D-Fender | devmod: Why would anyone not pick "yes" when they mean... well... YES? |
| 18:10.28 | p3nguin | Some people, when they mean yes, they say no. ;) |
| 18:10.54 | devmod | devmod: if videosupport is set to 'always' then RTP ports will be setup even when clients dont support it |
| 18:11.52 | [TK]D-Fender | devmod: And the point of that would be...? |
| 18:12.24 | devmod | its irrelevant to the point the feature does not work |
| 18:12.34 | devmod | thanks tho I found a bug filed regarding this issue |
| 18:12.43 | devmod | https://issues.asterisk.org/view.php?id=17016 |
| 18:12.59 | drmessano | Hmmm |
| 18:14.12 | beardy | Can one see which thread takes most resources/is causing problems, in some easy way? |
| 18:14.33 | *** join/#asterisk Yon (Yon@c80-217-244-16.bredband.comhem.se) |
| 18:14.56 | beardy | Can the identifier from "core show threads", 0xb710fb70 and so on, be connected to a thread PID, how? |
| 18:16.40 | *** part/#asterisk Scorcerer (scor@czlug.icis.pcz.pl) |
| 18:16.43 | [TK]D-Fender | devmod: What would it working accomplish? |
| 18:19.25 | devmod | [TK]D-Fender: Currently I have an end point that only supports audio enter a conference, then one that supports audio and video. When the third video enabled endpoint enter it only negotiates audio. I am not sure if that feature would solve the problem, but it was worth trying |
| 18:20.10 | drmessano | You can't three-way video |
| 18:20.29 | drmessano | Video can only be set up between two endpoints |
| 18:20.45 | [TK]D-Fender | ^^^ |
| 18:20.50 | [TK]D-Fender | devmod: * does not mix video |
| 18:20.54 | [TK]D-Fender | devmod: Never has |
| 18:21.35 | WIMPy | Maybe someone could just add video switching based on talk detection? |
| 18:22.10 | drmessano | Get someone to write the code.. you can add anything |
| 18:22.25 | drmessano | But as it stands now, that capability doesn't exist |
| 18:22.58 | [TK]D-Fender | wmipyNot happening. * is an audio switching platform, not multi-codec like that. Everything is always apssthrough and these apps have to be agnostic. |
| 18:23.11 | [TK]D-Fender | WIMPy: aLSO WHY WE ONLY HAVE AN 8KHZ MIXING CORE |
| 18:23.42 | devmod | [TK]D-Fender: I know I cant do mixing. I was expecting the two video enabled endpoints to see each other. The other member is audio only |
| 18:24.06 | drmessano | Doesn't matter.. the fat guy scares off the hot chicks, even if he does have a big wallet |
| 18:24.08 | WIMPy | [TK]D-Fender: Is the new bridging stuff also 8KHz only? |
| 18:24.21 | [TK]D-Fender | WIMPy: All of * is |
| 18:24.27 | paulc | Doesn't app_konference do something like that? or... maybe that's app_conference.. I know I read something about video switching based on active talkers somewhere.. |
| 18:25.06 | devmod | paulc: yes it does work but only when participants enter like this " videoenabled, videoenabled, audioonly" |
| 18:25.07 | [TK]D-Fender | devmod: the whole point in conferencing is the ends DON'T see each other. They are never in direct communication. * sits in between in conferences |
| 18:25.49 | devmod | paulc: if they enter like this "audioonly, videoenabled, videoenabled" then it does not work |
| 18:26.31 | drmessano | The bridging framework would make that interesting, but you need a full on application to have any control of the video |
| 18:26.51 | devmod | app_konference allows selection of videostream with dtmf |
| 18:27.20 | [TK]D-Fender | drmessano: Bridging only bridges the internal audio streams and assumes * is in the middle at all times, hence no adding video |
| 18:28.12 | devmod | [TK]D-Fender: same situation for video no? it is just a matter of selecting what stream to send to everybody ? |
| 18:28.38 | [TK]D-Fender | devmod: There is no "select" It's balls are cut off at the source. |
| 18:29.07 | devmod | [TK]D-Fender: anyway, since the order in which the participants entered affected the outcome, I thought maybe by using 'always' then endpoints would be always sending video no matter who else was in the konference |
| 18:29.18 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.178.236.dsl.dyn.forthnet.gr) |
| 18:29.46 | [TK]D-Fender | devmod: This is NOT happening. Pretty much end of discussion. |
| 18:30.01 | [TK]D-Fender | devmod: This would require a huge core re-write |
| 18:30.27 | devmod | [TK]D-Fender: I know it already has because I have tested a few products based on asterisk but ofc I don't have the resources to buy their solution |
| 18:30.47 | devmod | [TK]D-Fender: but thanks for the insight |
| 18:33.47 | *** join/#asterisk arielb27 (~chatzilla@63.214.236.169) |
| 18:34.05 | *** join/#asterisk cusco_ (~trilili@213.63.137.210) |
| 18:34.11 | cusco_ | hi |
| 18:34.18 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 18:34.25 | cusco_ | can directmedia=yes be used with IAX ??? |
| 18:38.09 | *** join/#asterisk gego (~gego@host-091-097-160-230.ewe-ip-backbone.de) |
| 18:39.15 | *** part/#asterisk gego (~gego@host-091-097-160-230.ewe-ip-backbone.de) |
| 18:39.59 | Katty | hi |
| 18:40.07 | Katty | has strawberries in 7up (= |
| 18:40.48 | cusco_ | I mean.. audio travels trough the same iax channel... |
| 18:40.54 | cusco_ | and its tcp/ip |
| 18:41.00 | cusco_ | so probably directmedia won't work |
| 18:41.01 | cusco_ | right? |
| 18:41.22 | WIMPy | udp off course |
| 18:41.36 | cusco_ | WIMPy: im refearing to IAX not SIP |
| 18:42.10 | WIMPy | Well, sip could be tcp, iax not. |
| 18:42.35 | cusco_ | ? |
| 18:42.48 | cusco_ | sip signalling is tcp, but audio is udp |
| 18:42.58 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 18:43.03 | cusco_ | while iax encapsulates both audio and signalling in the same tcp channel |
| 18:43.15 | *** join/#asterisk TimRiker (~timr@bzflag/projectlead/TimRiker) |
| 18:43.30 | WIMPy | no |
| 18:43.42 | cusco_ | err |
| 18:45.02 | cusco_ | where can I read mor about it then? |
| 18:45.57 | WIMPy | The channel.conf files? |
| 18:46.17 | *** join/#asterisk resno (~bryan@cpe-098-026-005-130.nc.res.rr.com) |
| 18:47.09 | cusco_ | WIMPy: ok I was just reading wikipedia on IAX |
| 18:47.32 | cusco_ | it seems that it opens a single UDP port for comm. |
| 18:47.36 | cusco_ | I was mistaken |
| 18:47.43 | cusco_ | WIMPy: does it support directmedia? |
| 18:48.20 | WIMPy | I *think* it always decides by itself. |
| 18:49.06 | resno | is it possible to use asterisk to make a home phone system for less then $10 a month USD? |
| 18:49.45 | ChannelZ | yes-ish, depending on how much you use the phone probably |
| 18:49.56 | WIMPy | resno: Depends on your electricity prices, I guess. |
| 18:49.59 | [TK]D-Fender | [14:47]<cusco_>it seems that it opens a single UDP port for comm. <- it DOES.. per direction |
| 18:50.17 | [TK]D-Fender | resno: * has nothing to do with the cost of SERVICE |
| 18:50.22 | *** join/#asterisk ldiamond (~chatzilla@24-226-250-193.tr.cgocable.ca) |
| 18:50.24 | resno | ChannelZ: its likley to sit then be used |
| 18:50.45 | [TK]D-Fender | resno: * is a telephony TOOLIKT, not a "service provider"" |
| 18:50.50 | [TK]D-Fender | toolkit* |
| 18:51.04 | ManxPower | ~toolkit |
| 18:51.05 | infobot | toolkit is, like, Remember Asterisk isn't really a PBX. Asterisk is a TOOLKIT that helps you build a PBX from scratch, much like libraries help you build an application from scratch. |
| 18:51.07 | resno | [TK]D-Fender: i understand, but i figure you would have service knowledge |
| 18:51.34 | [TK]D-Fender | resno: Depends what you expect to get for your $10/mo. You've been plenty vague on this... |
| 18:52.00 | [TK]D-Fender | ResI'll sell you 100 minutes a month for that. Sure. Yeppers |
| 18:52.25 | ManxPower | I'll sell you 200 mins for $10/month! |
| 18:52.28 | resno | [TK]D-Fender: ideally, i would like a phone number for outside access. but it wont be used very much. maybe 10-20 mins a month |
| 18:52.50 | [TK]D-Fender | resno: Then certainly. |
| 18:53.05 | [TK]D-Fender | resno: voip.me , les.net , bandwidth.con , etc |
| 18:53.10 | [TK]D-Fender | resno: Go shopping |
| 18:53.59 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 18:54.13 | resno | [TK]D-Fender: my problem is, i get lost in all the jargon. sip vs voip. |
| 18:54.32 | resno | some require two providers, etc |
| 18:54.44 | cusco_ | [TK]D-Fender: so is it possible to use directmedia=yes in iax? |
| 18:54.48 | [TK]D-Fender | resno: Some "what" require"two providers"? |
| 18:55.04 | [TK]D-Fender | resno: SIP *is* a VoIP protocol |
| 18:55.14 | resno | [TK]D-Fender: you need one provider for outbound and another for inbound |
| 18:55.17 | [TK]D-Fender | resno: Go read the BOOK... you need to know some of the basics of telephony |
| 18:55.19 | [TK]D-Fender | ~book |
| 18:55.20 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:56.03 | [TK]D-Fender | [14:55]<resno>[TK]D-Fender: you need one provider for outbound and another for inbound <- No. This is a CHOICE. Because most offer both services. Its a quesiton that some may be CHEAPER for certain parts making it WORTHWHILE to split your services up |
| 18:56.55 | resno | [TK]D-Fender: thanks for the link. ill read up and go from there |
| 18:58.45 | *** join/#asterisk italorossi (~italoross@201.76.154.130.intranet.digi.com.br) |
| 18:59.57 | ldiamond | I setup a SIP provider account and got a pbxes.org account, now I am trying to make a call but get some kind of recorded message which I can't really understand the first part, but ends with try again later. So I need to test out which part of the system is misconfigured / not working. Any suggestions? |
| 19:00.15 | ldiamond | But I should be able to make calls with a SIP provider and a pbxes.org account right? |
| 19:00.27 | ldiamond | Im using SIPdroid on my phone to make calls. |
| 19:01.07 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
| 19:02.09 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
| 19:04.32 | [TK]D-Fender | ldiamond: FreePBX is NOT supported here |
| 19:04.34 | [TK]D-Fender | ~freepbx |
| 19:04.35 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 19:04.53 | [TK]D-Fender | ldiamond: And yes whatever provider you chose should be able to work fine with *. |
| 19:05.10 | leifmadsen | or rather, Asterisk can work fine with practically any provider |
| 19:05.20 | leifmadsen | asterisk is the more flexible piece in that scenario :) |
| 19:05.25 | ManxPower | ~voipfd |
| 19:05.27 | infobot | extra, extra, read all about it, voipfd is http://ebooks-it.org/e-books/wiley/Wiley.VoIP.For.Dummies.Sep.2005.ISBN.0764588435.pdf |
| 19:11.59 | Infin1ty | i'm trying to register with broadworks sip trunk, i get timedout for some reason, though i can telnet the sip trunk |
| 19:11.59 | Nugget | telnet is eeeeeeevil! |
| 19:13.22 | p3nguin | You can telnet a "sip trunk," huh? |
| 19:14.23 | [TK]D-Fender | Infin1ty: Perhaps you should actually look at your call |
| 19:15.48 | Infin1ty | p3nguin, it was for saying that i can reach the trunk,, the port .. etc :\ |
| 19:16.17 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 19:17.02 | ManxPower | Infin1ty, You can't telnet to a SIP port unless you are one of the 0.001% of people that are running MS SIP Server |
| 19:17.14 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 19:17.20 | Infin1ty | sip turnk 5060 |
| 19:17.21 | ManxPower | telnet == tcp SIP == udp |
| 19:17.26 | Infin1ty | yes, i know, weird it worked :P |
| 19:17.29 | *** join/#asterisk Alagar (~Administr@122.164.36.154) |
| 19:17.32 | Infin1ty | hmm, i'll try netcat -u |
| 19:17.44 | ManxPower | Infin1ty, It is a sign of something seriously wrong |
| 19:17.58 | Infin1ty | well it works via udp, i can register to the sip with a softphone |
| 19:28.55 | *** join/#asterisk Qwell_ (~AndChat@pdpc/sponsor/digium/Qwell) |
| 19:29.00 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 19:29.01 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 19:47.52 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 19:50.19 | bent_screwdriver | if we're using a digium pri card with hw echo can, and a user is complaining about echo, wouldn't it show up in call recordings if it is echo that isn't being canceled by the VPM module? |
| 19:50.33 | ManxPower | bent_screwdriver, no |
| 19:50.34 | *** join/#asterisk hfb (~hfb@pool-98-112-146-69.lsanca.dsl-w.verizon.net) |
| 19:52.24 | ManxPower | bent_screwdriver, echo happens on ALL calls that have the far end being an analog line. This has nothing to do with VoIp. The thing is the echo happens so FAST you can't hear it. Just like you can't hear your voice bouncing off the walls when you are in a small room |
| 19:52.24 | ChannelZ | which end is complaining of the echo? |
| 19:52.37 | ManxPower | You start to HEAR echo when the latency gets high enough (the IP portion) and so you can't hear the echo in your recordings. |
| 19:54.30 | bent_screwdriver | ChannelZ: it is our end |
| 19:55.35 | ManxPower | bent_screwdriver, do you understand what I told you? |
| 19:56.36 | bent_screwdriver | ManxPower: these are Polycom IP650's hanging off switch with the PBX where it is directly connected and using QoS. |
| 19:57.52 | ManxPower | The latency I am referring to is part of ALL VoIP. |
| 19:58.04 | ManxPower | You should contact Digium for support on why your EC is not working. |
| 19:58.34 | QN | hi all. need a quick check. if there's any way that i can have a particular line of change made to extensions_additional.conf be saved permanent without it being overwritten everything i change something? |
| 19:58.47 | ManxPower | QN, Ask on #FreePBX |
| 19:58.50 | ManxPower | ~freepbx |
| 19:58.51 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 19:59.46 | *** join/#asterisk dajhorn (~dajhorn@206.16.96.160) |
| 19:59.51 | QN | thanks. |
| 20:00.02 | bent_screwdriver | is there a way with sip debug or something to see if there is still any latency |
| 20:01.03 | ManxPower | bent_screwdriver, You cannot get latency on VoIP low enough to not hear the echo. It is impossible. The only thing that can be done is to remove the ECHO that is caused by the FAR END ANALOG LINE, before you get the latency of VoIP i.e. on your PSTN interfface card. |
| 20:01.33 | ManxPower | The fact that your Digium EC is not working is something for Digium tech support. |
| 20:01.39 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 20:01.55 | bent_screwdriver | ManxPower: any experience in tweaking the VPM parameters, ie. vpmnlpmaxsupp, etc? |
| 20:02.33 | *** join/#asterisk helloritesh (~hellorite@121.243.172.188) |
| 20:02.41 | ManxPower | bent_screwdriver, What part of "you should contact Digium support" do you not understand. You can spend hours flapping your arms, trying different stuff, but it won't make any difference |
| 20:03.16 | ManxPower | Personally we use Sangoma cards -- never had EC problem with those. |
| 20:03.55 | helloritesh | guys, is there a way to specify two trunks in the Originate command as a failover mechanism. As in Dial command, the first one to answer connects. streamSock.send( "Channel: SIP/$phoneNum@trunk1,SIP/$phoneNum@trunk2\r\n", 0 ) |
| 20:03.58 | ManxPower | bent_screwdriver, Digium HW EC either works or it doesn't work. If it doesn't work then you need to contact Digium support. |
| 20:04.09 | ManxPower | ~siptrunk |
| 20:04.09 | infobot | extra, extra, read all about it, siptrunk is something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
| 20:04.19 | ManxPower | ~trunk |
| 20:04.20 | infobot | it has been said that trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 20:04.32 | bent_screwdriver | ManxPower: isn't this what this channel is for? I have a ticket open with Digium, dowgraded dahdi, tried alternate VPM firmware, etc. but they aren't helping so i am HERE..... |
| 20:04.48 | ManxPower | bent_screwdriver, this channel is to help with things we CAN help with. We can't help with EC |
| 20:04.49 | *** join/#asterisk Raden (~Raden@71.89.121.119) |
| 20:05.17 | p3nguin | I keep hearing bad things about Digium support. |
| 20:05.18 | ManxPower | If I was you I'd return the Dgium card and try a Sangoma |
| 20:05.19 | *** join/#asterisk Jumpie (n3rdz@ip68-230-28-186.ph.ph.cox.net) |
| 20:05.31 | bent_screwdriver | ManxPower: if anyone has experience in tweaking the vpm parameters then they COULD help |
| 20:05.38 | helloritesh | Ah! I did mention the trunk as well :( but my question is not related to trunk as such. |
| 20:05.56 | p3nguin | What was your question? |
| 20:06.00 | ManxPower | bent_screwdriver, but nobody has that experience here because we always have to contact Digium support for HW EC issues. |
| 20:06.06 | helloritesh | i want to be able to dial out via 2 trunks and le the first one that answers win |
| 20:06.14 | helloritesh | much the same as dial command allows |
| 20:06.18 | helloritesh | but via AMI |
| 20:06.23 | helloritesh | using Originate |
| 20:06.29 | bent_screwdriver | ManxPower: 10-4. Do you have better experiene with Sangoma? |
| 20:06.32 | ManxPower | helloritesh, I wish you the BEST of luck |
| 20:06.37 | helloritesh | <PROTECTED> |
| 20:06.43 | p3nguin | Now I understand why ~trunk was displayed... but were you going to rephrase your question to make it make more sense? |
| 20:06.45 | helloritesh | really? |
| 20:06.48 | ManxPower | bent_screwdriver, we have never had an echo problem with Sangoma |
| 20:06.49 | helloritesh | it's that bad? |
| 20:06.54 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 20:07.05 | helloritesh | should i try this out? |
| 20:07.25 | ManxPower | In the Dialplan using the Dial command it would be Dial(SIP/device1&SIP/device2) |
| 20:07.35 | helloritesh | yeah |
| 20:07.36 | bent_screwdriver | ManxPower: I bouught digium support so I was afraid if i strayed with the hardware, and there was an issue, digium would always point back to that....worth testing i guess. |
| 20:07.39 | helloritesh | sorry |
| 20:07.43 | helloritesh | <PROTECTED> |
| 20:07.51 | [TK]D-Fender | helloritesh: Dial a local channel that dials out both |
| 20:08.06 | p3nguin | You could dial two devices in sequence; if the first one doesn't answer, the second one might. |
| 20:08.06 | helloritesh | Gotcha |
| 20:08.07 | helloritesh | clever |
| 20:08.38 | ManxPower | bent_screwdriver, As much as I love Sangoma, it would be stupid to not use the support you are entitled to from Digium |
| 20:08.47 | p3nguin | Except when they aren't providing said support... |
| 20:08.48 | helloritesh | i wanted the trunk failover .. not so much as the findme followme.. type.. So [TK]D-Fender |
| 20:08.52 | helloritesh | s trick might work |
| 20:08.57 | p3nguin | at which time a return of merchandise is in order. |
| 20:09.15 | ManxPower | If Digium cannot solve the issue then I have no hesitation on recommending Sangoma. They are harder and more confusing to configure, but they work very well |
| 20:09.24 | helloritesh | Great.. Thanks a bunch guys.. |
| 20:10.32 | bent_screwdriver | ManxPower: thanks. i'll continue to work through this support ticket and see what happens |
| 20:10.33 | *** join/#asterisk russellb (~russellb@asterisk/digium-open-source-team-lead/russellb) |
| 20:10.34 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:12.08 | ManxPower | bent_screwdriver, Digium has some kind of "total satisfaction quarantine" |
| 20:12.16 | ManxPower | LOL! So much for spell check |
| 20:12.34 | ManxPower | <PROTECTED> |
| 20:12.41 | p3nguin | lol |
| 20:14.58 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 20:15.00 | emora | Today I ran into an issue with Dial(SIP/device1&SIP/device2) perhaps someone could help me understand what happend |
| 20:15.52 | emora | the first number in the dial gets dialed but the second shows an error on the console |
| 20:16.03 | emora | it just says "channel unavailable" |
| 20:17.09 | *** join/#asterisk Qwell_ (~AndChat@pdpc/sponsor/digium/Qwell) |
| 20:17.09 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 20:17.10 | *** join/#asterisk grungies1138 (~grungies1@cpe-66-74-197-34.san.res.rr.com) |
| 20:17.14 | grungies1138 | hello |
| 20:17.16 | emora | I checked the syntax. Everything looked correct. I tested each of the numbers separately ... they're both ok |
| 20:18.06 | emora | I wasnt able to spent much time on it, but I wasnt able to figure it out. Any clues? |
| 20:18.13 | ManxPower | emora, put a copy of the CLI output on pastebin.ca |
| 20:25.01 | emora | ManxPower: I'm trying to connect to that server and reproduce the message |
| 20:26.37 | [TK]D-Fender | <yoda> Do or do no. There is no try. |
| 20:27.09 | emora | Got it. Its not chan unavail ... it says "cause 20 - Unknown" - I'll clean it up and drop into pastebin |
| 20:27.45 | *** join/#asterisk TimeRider (steve@5ac7b358.bb.sky.com) |
| 20:27.59 | [TK]D-Fender | checkout time, BBIAB |
| 20:28.03 | *** join/#asterisk bn-7bc (~bjarne-im@213.236.252.25) |
| 20:28.11 | bn-7bc | =j #cisco |
| 20:28.36 | Jumpie | fail |
| 20:28.38 | bn-7bc | pardon the typo / turd into = |
| 20:28.51 | *** join/#asterisk afink (~afink@204.26.87.226) |
| 20:29.35 | Katty | omnomnomnoms babycarrots |
| 20:29.46 | emora | here it is http://pastebin.com/nmESYKiz |
| 20:30.49 | Jumpie | who's babycarrots?! |
| 20:30.51 | Jumpie | lol |
| 20:31.40 | Katty | mine. |
| 20:31.42 | Katty | clearly. |
| 20:32.13 | bmoraca_work | baby carrots are excellent with ranch dressing |
| 20:32.19 | bmoraca_work | even better with french dressing |
| 20:32.30 | *** join/#asterisk DrDamnit (~michael@173-165-161-161-atlanta.hfc.comcastbusiness.net) |
| 20:32.49 | Katty | not a big fan of ranch, never tried them with french before |
| 20:33.01 | Jumpie | i love sugar snap peas |
| 20:35.45 | ManxPower | emora, you do not have [1234015] section in your sip.conf |
| 20:36.05 | *** join/#asterisk Raden (~Raden@71.89.121.119) |
| 20:36.33 | ManxPower | because of that Asterisk thinks you are dialing the IP address 1234015 |
| 20:36.50 | emora | got it! |
| 20:37.56 | *** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no) |
| 20:38.15 | ManxPower | *whine* someone install this WiFi AP at the top of this 24' telephone pole for me. *whine* |
| 20:38.27 | Qwell_ | Katty |
| 20:38.32 | Katty | mister Qwell |
| 20:38.34 | Katty | with a tail. |
| 20:38.34 | ManxPower | She would be perfect! |
| 20:38.40 | Qwell_ | Amuse me! |
| 20:38.54 | Katty | pours Qwell a drink of strawberries and 7up |
| 20:39.13 | emora | ManxPower: Thank you |
| 20:39.26 | Qwell_ | Can't eat/drink |
| 20:39.48 | Katty | why |
| 20:40.27 | ManxPower | Katty, He;s a vampire |
| 20:40.30 | Katty | do you have a test in the morning |
| 20:40.59 | Kobaz | it's katty! |
| 20:41.11 | Katty | shh don't tell |
| 20:41.36 | grungies1138 | Hey guys, I have an issue with outbound calls on a SIP Trunk. |
| 20:41.41 | Qwell_ | See twitter! |
| 20:41.46 | ManxPower | ~ask |
| 20:41.55 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 20:42.19 | ManxPower | grungies1138, Are you using FreePBX or Trixbox or other GUI add-on for Asterisk? |
| 20:42.36 | Kobaz | i added a comment |
| 20:42.39 | Kobaz | er |
| 20:43.17 | grungies1138 | I'm registered but when I try to dial out, it tells me the : Call from '208' to extension '91XXXXXXXXXX' rejected because extension not found. |
| 20:43.27 | grungies1138 | straight up Asterisk |
| 20:43.41 | paulc | do you have a matching 91XXXXXXXXXXXXx extension in the right/relevant context? |
| 20:44.10 | grungies1138 | I added a context called [outgoing_calls] |
| 20:44.22 | *** join/#asterisk bmoraca (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 20:44.23 | bmoraca | uhg |
| 20:44.50 | grungies1138 | I included it in my [users] context |
| 20:44.59 | afink | grungies1138: show us the dial plan |
| 20:45.04 | afink | ~paste |
| 20:45.05 | infobot | i guess paste is http://rafb.net/paste/, or see also pb, or http://bin.cakephp.org/ |
| 20:45.06 | paulc | so does [outgoing_calls] have an extension that matches 91XXXXXXXXXXX ? |
| 20:45.26 | grungies1138 | let me pastebin my extensions.conf and sip.conf |
| 20:45.28 | grungies1138 | one moment |
| 20:46.21 | *** join/#asterisk Infin1ty (~Infin1ty@pdpc/supporter/active/infin1ty) |
| 20:46.59 | *** join/#asterisk LeandroAugusto (~leandro.a@189.90.54.130) |
| 20:48.41 | grungies1138 | http://pastebin.com/Jv1qUSbz |
| 20:49.16 | grungies1138 | other than the register => everything else is default |
| 20:51.25 | p3nguin | grungies1138: You don't have an extension matching 91anything. |
| 20:51.37 | p3nguin | That's why it isn't found. |
| 20:51.57 | afink | I also don't see ${TRUNK} defined anywhere |
| 20:52.14 | grungies1138 | but shouldn't _9NXXNXXXXXX do that? |
| 20:52.17 | p3nguin | For now, the context/extension is hte problem. |
| 20:52.23 | p3nguin | grungies1138: no. |
| 20:52.39 | grungies1138 | then how should it be? |
| 20:52.40 | p3nguin | _9N does not match 91. |
| 20:52.50 | p3nguin | N = 2-9 |
| 20:52.54 | grungies1138 | ahh! |
| 20:52.59 | *** join/#asterisk dgd (~user@195.230.115.10) |
| 20:53.10 | afink | _91NXXNXXXXXX |
| 20:53.33 | ldiamond | I have setup an account with a SIP provider, I have a pbx setup, and I have a DID. I can make outbound calls, but inbound calls won't reach my phone. |
| 20:53.59 | p3nguin | ldiamond: Put your sip debug in pastebin.com. |
| 20:54.03 | ldiamond | I see in the log that my pbx gets the call, but it just hangs up and sounds like a fast busy signal |
| 20:54.15 | p3nguin | ldiamond: Also put your extensions.conf in a separate paste. |
| 20:55.17 | ldiamond | sip debug? |
| 20:55.46 | ldiamond | ~pastebin |
| 20:55.47 | infobot | [~pastebin] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 20:55.47 | p3nguin | If you want help on a SIP problem, you have to provide sip debug output. |
| 20:56.16 | ldiamond | http://asterisk.pastey.net/136928 thats the ext.conf |
| 20:56.49 | grungies1138 | ok, I fixed the _91NXXNXXXXXXm defined $TRUNK as SIP/<sip_provider_context> |
| 20:57.02 | p3nguin | If THAT is your extensions.conf, I wouldn't expect ANY calls to work. |
| 20:57.21 | p3nguin | You don't have globals nor any other contexts. |
| 20:57.42 | afink | grungies1138: work? |
| 20:57.46 | grungies1138 | nope |
| 20:57.51 | grungies1138 | same error |
| 20:57.55 | afink | reload? |
| 20:57.58 | grungies1138 | say wait... |
| 20:57.58 | p3nguin | which error? |
| 20:57.59 | grungies1138 | yeah |
| 20:58.26 | grungies1138 | [May 25 13:58:15] NOTICE[851]: chan_sip.c:20039 handle_request_invite: Call from '208' to extension '917603301708' rejected because extension not found. |
| 20:58.58 | ldiamond | p3nguin: I guess this is because my pbx is from pbxes.org |
| 20:59.32 | p3nguin | '917603301708' should be matched by _91NXXNXXXXXX |
| 21:00.03 | p3nguin | grungies1138: More info is needed. |
| 21:00.16 | grungies1138 | should my [users] and [outgoing_calls] contexts be included anywhere? like default or something? |
| 21:00.43 | p3nguin | What is the context for the peer [208] from sip.conf? |
| 21:00.59 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 21:01.22 | grungies1138 | [208] |
| 21:01.40 | p3nguin | That's the peer name, now try giving the answer to what I asked. |
| 21:02.10 | grungies1138 | oh. so I can add context=users under 208? |
| 21:02.24 | afink | you already have it on your paste |
| 21:02.33 | p3nguin | You have to set a context for the call to start in. |
| 21:02.45 | p3nguin | If you don't define it, it would be the default global context. |
| 21:02.47 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:02.56 | p3nguin | which is bad. |
| 21:03.09 | p3nguin | If you set context=users, then the call would start in [users]. |
| 21:03.19 | p3nguin | Then it could be reasonable to do "include => outgoing_calls" in [users] (in extensions.conf). |
| 21:03.35 | *** join/#asterisk Janos (~cramos@190.10.52.113) |
| 21:03.48 | p3nguin | And that, boys and girls, is how the dialplan hierarchy is created. |
| 21:04.32 | KavanS | p3nguin, yes - I wish we had done things the hard way instead of beginning with asterisk-gui |
| 21:04.50 | p3nguin | This is the easy way. |
| 21:04.55 | KavanS | def. worth the effort to say the least - to learn the knowledge, or hire out digium or a consultant :) |
| 21:05.56 | p3nguin | I tried the asterisk-gui once. It failed miserably. |
| 21:06.34 | idespinner | p3nguin, i'm guessing you never had to deal with digiums AA50. Support mandated GUI only |
| 21:06.52 | idespinner | and that was the early version of asterisk-gui |
| 21:07.16 | p3nguin | It probably worked better than the gui works with current asterisk versions. |
| 21:07.16 | grungies1138 | I did do that |
| 21:07.35 | idespinner | it was the same version with only appliance specific tweaks |
| 21:07.40 | [TK]D-Fender | idespinner: Hence a great reason to never have bought one :) |
| 21:07.50 | p3nguin | grungies1138: save your files. Now run "sip reload" and "dialplan reload" and try your call again. |
| 21:08.30 | [TK]D-Fender | And the fact that Asterisk-gui isn't even maintained right now an even better one. |
| 21:09.55 | grungies1138 | getting a different error now. but I'll have to troubleshoot that later. gotta go pick up my son |
| 21:09.55 | *** join/#asterisk DarkRift (~dark@modemcable015.68-200-24.mc.videotron.ca) |
| 21:23.04 | *** join/#asterisk markit (~marco@88-149-177-66.static.ngi.it) |
| 21:33.41 | *** join/#asterisk scalex000 (~chatzilla@190.167.9.147) |
| 21:33.49 | scalex000 | good afternoon, |
| 21:33.56 | scalex000 | Hi everyone |
| 21:34.50 | scalex000 | what is the variable name to know which extesion is dialing |
| 21:36.35 | *** join/#asterisk bjhaid (~IceChat7@41.220.68.3) |
| 21:37.15 | p3nguin | ${EXTEN} |
| 21:43.37 | shader | is there a way besides ${CALLERID(num)} to figure out who's calling? I want to do something based on the local extension number, even if they've set a different caller id |
| 21:44.50 | [TK]D-Fender | shader: Stop calling DEVICES "extensions" |
| 21:45.08 | [TK]D-Fender | shader: Look at the CHANNEL name |
| 21:48.58 | *** join/#asterisk KNERD (~KNERD@129.113.44.106) |
| 21:51.24 | *** join/#asterisk niekvlessert (~niek@82-171-252-6.ip.telfort.nl) |
| 21:51.48 | niekvlessert | hello! i'm using bri cards and sometimes there's echo and sometimes calls will break off |
| 21:51.58 | niekvlessert | maybe timing? |
| 21:52.08 | niekvlessert | how would I find out about timing issues? |
| 21:52.43 | niekvlessert | and what timing issues can there be? Can it help if I change the primary timing source? |
| 21:53.31 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 21:54.28 | niekvlessert | what does this do? Set option TONE VERIFY, mode: MUTECONF(1) |
| 21:55.41 | ldiamond | What is a good sip client for PC (so I can call using my SIP provider)? |
| 21:55.53 | niekvlessert | zoiper / x-lite |
| 21:57.04 | *** join/#asterisk Yon (Yon@c80-217-244-16.bredband.comhem.se) |
| 21:58.29 | *** join/#asterisk bjhaid (~IceChat7@41.220.68.2) |
| 21:59.45 | niekvlessert | is this acceptable with zttest?: Best: 99.999 -- Worst: 99.899 -- Average: 99.960807, Difference: 99.996731 |
| 22:00.31 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 22:02.06 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 22:02.44 | *** join/#asterisk shader (~40846872@gateway/web/freenode/x-lnhpmdspxkzephvq) |
| 22:02.49 | shader | sorry, my client bugged |
| 22:02.51 | *** join/#asterisk QubeZ (~nkasu@fl-71-1-80-37.sta.embarqhsd.net) |
| 22:02.52 | QubeZ | hello all |
| 22:03.05 | shader | did anyone see/answer my question, or should I repeat it? |
| 22:04.30 | KNERD | repeat it |
| 22:05.08 | shader | I was wondering what the best way to figure out the extension number of the calling party, even if they have a caller id # set in sip.conf |
| 22:05.31 | shader | this is for local phones, to determine things like call forwarding, voicemail, etc. |
| 22:05.37 | [TK]D-Fender | [17:43]<shader>is there a way besides ${CALLERID(num)} to figure out who's calling? I want to do something based on the local extension number, even if they've set a different caller id |
| 22:05.39 | [TK]D-Fender | [17:44]<[TK]D-Fender>shader: Stop calling DEVICES "extensions" |
| 22:05.41 | [TK]D-Fender | [17:45]<[TK]D-Fender>shader: Look at the CHANNEL name |
| 22:06.13 | shader | thank you |
| 22:08.31 | *** join/#asterisk bjhaid (~IceChat7@41.220.68.1) |
| 22:10.42 | bmoraca_work | stupid CUE script editor won't launch |
| 22:10.45 | bmoraca_work | piece of crap |
| 22:11.10 | shader | so how would you do that? ID=${Cut(${CHANNEL},'/',2)} ? |
| 22:11.56 | [TK]D-Fender | shader: You might want to consider trying |
| 22:12.33 | bmoraca_work | but it's just so much easier to ask you to do it for him! |
| 22:14.38 | diegomad | when making a originate in the device displays caller = asterisk, this can be changed? |
| 22:15.31 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 22:15.34 | [TK]D-Fender | diegomad: CLI Originate no, AMI, yes |
| 22:15.54 | [TK]D-Fender | diegomad: Unless the channel you originate is a LOCAL one at which point you can do whatever the hell you want |
| 22:16.30 | diegomad | [TK]D-Fender, thanks :D |
| 22:24.02 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:27.03 | shader | is it possible to do variable processing on a channel if it's in a context, before an extension has been picked? |
| 22:27.46 | *** join/#asterisk Sipster_ (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 22:27.47 | p3nguin | The extension is where the processing starts. |
| 22:27.57 | shader | ok |
| 22:28.18 | scalex000 | shader, thx, I found it |
| 22:28.29 | *** join/#asterisk Qwell_ (~AndChat@pdpc/sponsor/digium/Qwell) |
| 22:28.29 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 22:28.31 | *** join/#asterisk techie (~root@unaffiliated/techie) |
| 22:29.00 | *** join/#asterisk mindCrime (~chatzilla@cpe-075-189-213-049.nc.res.rr.com) |
| 22:29.05 | [TK]D-Fender | shader: You aren't in a context if you haven't got an extension |
| 22:29.25 | scalex000 | TK, to setup a ODBC connection on Asterisk only just use odbc.ini and odbcinst.ini and func_odbc.conf |
| 22:29.44 | [TK]D-Fender | scalex000: If you want to go that way... sure. |
| 22:30.09 | shader | [TK]D-Fender: Ok, I was just wondering if there was a way to do generalized preprocessing without creating another context and two GoTos |
| 22:30.28 | scalex000 | TK, I get access denied, I put the right access |
| 22:30.40 | scalex000 | TK, I restart asterisk |
| 22:31.20 | [TK]D-Fender | scalex000: Access denied means what it says. It means it got to ask and the other side said "no" |
| 22:31.36 | [TK]D-Fender | shader: There is no "pre"-process |
| 22:31.47 | [TK]D-Fender | shader: There is only "process" |
| 22:32.05 | p3nguin | and it starts when you make a call. |
| 22:42.58 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 22:44.06 | shader | in what cases should I have asterisk play ring sounds, and when is it likely to be handled by the other end? |
| 22:46.16 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 22:46.24 | p3nguin | You almost never need to play a ringing sound. |
| 22:46.39 | p3nguin | If you do, you'll already know why you're going to do it. |
| 22:51.19 | *** part/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 22:52.41 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 22:53.52 | shader | ok, that's weird. Apparently it's not a lack of ringtone, but strange intermittent behavior |
| 22:57.05 | shader | any idea why sometimes it "is making progress" so it passes connects the call back to the channel, but other times it doesn't? |
| 22:59.53 | shader | http://pastebin.com/EaaiYdgP |
| 23:01.43 | *** join/#asterisk Raden (~Raden@71.89.121.119) |
| 23:02.04 | [TK]D-Fender | ShaIn neither case do we see the complete call end to en, and no SIP DEBUG |
| 23:02.29 | shader | sip debug? |
| 23:04.03 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 23:04.53 | bmoraca_work | woot |
| 23:05.01 | bmoraca_work | finally got CUE script editor to load... |
| 23:05.06 | bmoraca_work | ....in a Windows 98 virtual machine |
| 23:05.19 | bmoraca_work | kind of, lol |
| 23:05.52 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 23:05.58 | *** join/#asterisk jks (jks@193.189.93.254) |
| 23:10.45 | Infin1ty | i have a problem with a sip trunk, it does not send a authusername although it is set in the string |
| 23:11.40 | shader | [TK]D-Fender: this sip debug stuff outputs a TON of stuff, you sure you want to see all of it? |
| 23:11.58 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 23:12.34 | *** join/#asterisk SaiSoma|AtHome (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 23:13.30 | bmoraca_work | man...CUE Script Editor makes me miss Asterisk dialplan code something FIERCE |
| 23:16.44 | shader | wow |
| 23:18.48 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 23:19.00 | shader | have an example? |
| 23:20.33 | shader | or is it mostly the editor, instead of the script? |
| 23:20.44 | bmoraca_work | mostly the editor |
| 23:20.47 | bmoraca_work | it's aweful |
| 23:21.03 | bmoraca_work | and it's not really much of a "script" either... |
| 23:21.06 | bmoraca_work | it's aweful, too |
| 23:21.23 | shader | hmm |
| 23:21.41 | shader | must be pretty bad if you're pining for asterisk dialplan code :P |
| 23:22.07 | bmoraca_work | asterisk dialplan isn't that bad. it's pretty powerful and I rather enjoy working with it |
| 23:22.47 | ldiamond | Can anyone suggest good free softphones? |
| 23:23.09 | shader | for linux, Twinkle |
| 23:23.32 | shader | for everyone else X-lite is popular |
| 23:23.51 | shader | can't really vouch for good, as I haven't used it much |
| 23:24.49 | ldiamond | anything else for windows? |
| 23:25.36 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 23:27.53 | shader | I haven't used anything else for windows personally |
| 23:32.06 | bmoraca_work | holy crap |
| 23:32.23 | bmoraca_work | 30-year FHA home loans are at 4.75% |
| 23:32.26 | bmoraca_work | insanity |
| 23:33.15 | KavanS | any guide out there - that can explain how EU phone systems differ from U.S. ? - i.e. what the differences in the numbers dialed mean etc. |
| 23:33.25 | KavanS | I am not sure if I can identify area/region codes from this long ass number, and am confused |
| 23:40.25 | *** join/#asterisk Carlos_Tico (~carlos@c-98-201-56-25.hsd1.tx.comcast.net) |
| 23:41.24 | *** join/#asterisk eskaypey (~eskaypey@unaffiliated/eskaypey) |
| 23:41.31 | eskaypey | greetings |
| 23:41.55 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 23:42.14 | eskaypey | i've got a more of a general pbx question, if i may, when i unplug a voip phone and plug it back in it doesn't come back on |
| 23:42.59 | eskaypey | is it because the pbx doesn't know it is there yet? can i force it to detect it? |
| 23:44.09 | bmoraca_work | that question makes no sense |
| 23:44.39 | paulc | KavanS: wikipedia has a list of country codes - if your number is in E.164 format, you can figure out the country code by the first 1 to 3 digits of the number. After that, figuring out region/city etc gets tricky.. or trickier at least.. |
| 23:45.08 | paulc | eskaypey: unplug the power? or the network cable? plug the network cable into a different port? can you elaborate on your question |
| 23:45.16 | KavanS | paulc, ok roger that - learning how to dial out our phone system to EU area codes - def. different from u.s. :P |
| 23:45.53 | eskaypey | paulc: sorry, the phone uses PoE, so its got just one cat5 cable going into it |
| 23:46.37 | eskaypey | the phone would be working normally, then i unplug it, say to test another phone, after that plug my phone back in and it doesn't boot up |
| 23:47.17 | paulc | KavanS: Yeah, some might argue it's a bit more sensible in Europe ;-) Generally 0 = long distance, 00 = international.. and the nice thing in europe is you can generally dial full national number even if it's a local call. US PSTN barfs at that. |
| 23:47.52 | paulc | eskaypey: "doesn't boot up" as in.. download configs, successfull register etc? |
| 23:48.09 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 23:49.24 | eskaypey | paulc: yep |
| 23:49.30 | KavanS | paulc, yeah it makes "some" sense, but am trying to dial local numbers and it's giving me a headache - I suppose I need access to this other asterisk system to really troubleshoot :P |
| 23:49.53 | *** join/#asterisk korihor (~korihor@190.205.240.212) |
| 23:52.33 | paulc | KavanS: where is it, and where are you? |
| 23:52.46 | paulc | eskaypey: What kind of phone? |
| 23:54.55 | eskaypey | paulc: Aastra OpenPhone |
| 23:55.40 | eskaypey | paulc: have a suspicion it happens because its going through the switch? would the switch play up with it? |
| 23:55.54 | eskaypey | doubt it.. |